With --gapless-audio=no, changing from one file to the next apparently
made it hang, until the player was woken up by unrelated events like
input. The reason was that the AO doesn't notify the player of EOF
properly. the played was querying ao_eof_reached(), and then just went
to sleep, without anything waking it up.
Make it event-based: the AO wakes up the playloop if the EOF state
changes.
We could have fixed this in a simpler way by synchronously draining the
AO in these cases. But I think proper event handling is preferable.
Fixes: #1069
CC: @mpv-player/stable (perhaps)
Somehow, there was a larger misunderstanding in the code: ao_buffer
does not need to be preserved over audio reinit for proper support of
gapless audio. The actual AO internal buffer takes care of this.
In fact, preserving ao_buffer just breaks audio resync. In the ordered
chapter case, end_pts is used, which means not all audio data in the
buffer is played, thus some data is left over when audio decoding
resumes on the next segment. This triggers some code that aborts resync
if there's "audio decoded" (ao_buffer contains something), but no PTS
is known (nothing was actually decoded yet).
Simplify, and always bind the output buffer to the decoder.
CC: @mpv-player/stable (maybe)
The oldest supported FFmpeg release doesn't provide
av_vdpau_alloc_context(). With these versions, the application has no
other choice than to hard code the size of AVVDPAUContext. (On the other
hand, there's av_alloc_vdpaucontext(), which does the same thing, but is
FFmpeg specific - not sure if it was available early enough, so I'm not
touching it.)
Newer FFmpeg and Libav releases require you to call this function, for
ABI compatibility reasons. It's the typcal lakc of foresight that make
FFmpeg APIs terrible. mpv successfully pretended that this crap didn't
exist (ABI compat. is near impossible to reach anyway) - but it appears
newer developments in Libav change the function from initializing the
struct with all-zeros to something else, and mpv vdpau decoding would
stop working as soon as this new work is relewased.
So, add a configure test (sigh).
CC: @mpv-player/stable
Don't worry, your ~/.config/... paths are safe. This merely removes
handling of $XDG_CONFIG_DIRS for global paths.
Maybe there is a better solution for this, like still including the
"traditional" config dir. But I will leave the fine reading of this
(crappy) spec and fixing the code accordingly to someone else. So, if
anyone has interest in getting this behavior back, you will have to
write a patch. This patch should _also_ not break expected behavior.
Fixes#1060.
Loading a playlist with --playlist from a sub-directory added the
playlist's base path twice: one time in the playlist demuxer, and then
again in playlist_parse_file(). The latter function is used only for
--playlist, so it worked when loading the playlist directly.
(This is probably a mess-up when the MPlayer playlist parsers were
replaced with newer code.)
CC: @mpv-player/stable
The audio/video sync code in player/audio.c calls ao_reset() each time
audio decoding is entered, but the player is paused, and there would be
more than 1 sample to skip to make audio start match with video start.
This caused a wakeup feedback loop with push.c.
CC: @mpv-player/stable
When embedding a X window, it's hard to control whether it receives
mouse/keyboard input or not. It seems the X protocol itself makes this
hard (basically due to the outdated design mismatching with modern
toolkits), and we have to take care of these things explicitly.
Simply do this by manually querying and using the parent window event
flags.
This restores some MPlayer behavior (it doesn't add back exactly the
same code, but it's very similar).
This probably has some potential to interfere with libmpv embedding, so
bump the client API minor.
CC: @mpv-player/stable (if applied, client-api-changes.rst has to be
adjusted to include the 0.5.2 release)
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
Conflicts:
player/playloop.c
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
The little lua snippet at #488 as well as the actual implementation
seems to indicate that not expanding properties is indeed the correct
behavior. Document that.
Signed-off-by: wm4 <wm4@nowhere>
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)
The previous commit made the completion script always return non-zero, even when
a match is found. This explicitly sets the return value to zero whenever a match
is found but defaults to non-zero in case nothing is matched.
Returning a non-zero value signals to the zsh completion system that no matches
were added by the script so that it can try the user-defined matchers (e.g.
those defined with matcher-list).
Fixes#1008.
Otherwise vdp_video_mixer_destroy() would later fail when called on an invalid
video mixer handle. With mesa r600 vdpau driver, this would cause a segfault.
Found with valgrind. This is somewhat terrifying, because the VA-API API
function is supposed to fill these values, and we access them only if
the API functions return success. So this shouldn't have happened.
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.)
When a new event was added, merely a flag was set, instead of actually
waking up the core (if needed). This was ok in ancient times when all
event sources were part of the select() loop. But now there are several
cases where other threads can add input, and then you actually need to
wakeup the core in order to make it read the events at all.
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.
vo_sdl.c has broken event handling and just polls. The polling time was
quite low, so the playloop OSD redrawing heuristic inhibited redraws,
which made the window appear frozen when paused.
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.
Some of them changed semantics or got renamed.
Note that the replacements in the example.conf are not necessarily the
equivalents of the replaced options.
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.)
Split the options into the following sections:
* Playback Control
* Program Behaviour
* Video
* Audio
* Subtitles
* Window
* Disc Devices
* Equalizer
* Demuxer
* Input
* OSD
* Screenshot
* Software Scaler
* Terminal
* TV
* Cache
* Network
* DVB
* PVR
* Miscellaneous
Most options are sorted by usefullness and how often they're used or how
important they are.
This makes finding the right options easier and adds some sort of structure.
Reduce from 1000ms to 100ms. Since there is an audio thread updating AOs
quickly enough now, requesting such a large buffer size makes no sense
anymore.
Changing audio-delay is probably not needed.
Changing balance "works", but not as expected (sets up a pan matrix to
change left/right contributions to each other, rather than changing the
relative volumes of each channel).
I expect that the rest are not in use by anyone.
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.
Switching tracks caused cached_demux_control() to catch the command to
switch tracks, even if no thread was running. Thus, the tracks were
never really switched, and EOF happened immediately on playback start.
Fix it by not using the cache at all if the demuxer thread is disabled.
The cache code still has to be called somewhere, though, because it
handles stream metadata update.
Regression from today.
Got lost some time ago. Although I'm not sure if it actually does
anything on MPlayer (it exists there, but might broken, not sure).
Fixes#988 (untested).
This can just happen in the time between VO creation, and the first call
to vo_reconfig. It seems the recent threading changes exposed this bug.
Fixes#986.
Sometimes GetClientRect() appeared to fail during init, and since we
don't check GetClientRect() calls (because they're on our own window,
and logically can never fail), bogus resizes were triggered. This could
cause vo_direct3d to fail initialization.
The reason was that w32->window was set to 0 during early window
initialization: CreateWindow*() can send messages to the new window,
even though it hasn't returned yet. This means w32->window is not yet
set to our window handle, and functions in WndProc may accidentally pass
hwnd=0 to win32 API functions.
Fix it by initializing w32->window on opportunity. This also means we
always strictly expect that the WndProc is used with our own window
only.
Because why not.
This can lead to reordering of operations between seeking and track
switching (happens when the demuxer wakes up after seek and track
switching operations were queued). Do the track switching strictly
before seeks if there is a chance of reordering, which guarantees that
the seek position will always start with key frames. The reverse
(seeking, then switching) does not really have any advantages.
(Not sure if the player relies on this behavior.)
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.
This fixes the fullscreen issues on Intel for me. I'm baffled by it and
don't understand why this suddenly works. Intel drivers being shit?
Windows being shit? HWND or HDC being only 97% thread-safe instead of
98%? Me missing something subtle that is not documented anywhere?
Who knows.
Now instead of creating the HDC and OpenGL context on the renderer
thread, they're created on the GUI thread. The renderer thread will
then only call wglMakeCurrent, SwapBuffers, and OpenGL standard
functions.
Probably fixes github issue #968.
We avoided reindenting this in the past to allow merging upstream changes.
In hindsight these are very unlikely and we are actually doing changes on
the code, so it's better to have the correct indentation and formatting in
our source file.
The parser can be called with dst (the target) set to NULL if the option
should be verified only. The code didn't respect this, and could result
in crashes when used in config profiles or filter sub-options.
Fixes#981.
This is probably nicer. The actual version number doesn't change (other
than the minor being incremented).
The "| 0UL" is to make the type unsigned long int, like it was before.
Instead of using a regex to match names to be exported from the libmpv
dynamic shared library, use a libmpv.def file, which lists all exported
functions explicitly.
This reduces the platform specifics in syms.py. I'm not sure if the
separate compile_sym task is still needed (it could probably be
collapsed, which would concentrate the platform specifics into one
place).
libmpv requires nm for creating the list of exported symbols (this is
done in syms.py). We should probably just make this list static instead,
but since this involves platform-specific code, for now this quick-fix
will do.
It seems the vdpau API does not support these.
Do a semi-expensive emulation of it. On the other hand, it's not like
this is a commonly-used feature. (It might be better to make --vf=flip
always copy instead of flipping it via pointer tricks - but everything
allows flipped images, and even decoders or libavfilter filters could
output them.)
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.
If the compositor sends a configure event immediately after a window is
created (for example, if it implements tiling window management), mpv
will attempt to call wl_egl_window_resize before it has actually created
the egl_window, causing a crash.
Nobody uses 'c' (except accidentally) - remove.
Everyone agrees that OSD level cycling on 'o' is dumb, so map it to
show_progress instead. Cycling the OSD level is now available on 'O'.
No reason to ummap 'P' yet.
Also see issue #973.
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.)
Use OPT_KEYVALUELIST() for all places where AVOptions are directly set
from mpv command line options. This allows escaping values, better
diagnostics (also no more "pal"), and somehow reduces code size.
Remove the old crappy option parser (av_opts.c).
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...)
If OpenGL 3.x doesn't work, the fallback to legacy GL will call the
function to create the DC again, and it will assert. Just make it not
to, and also simplify the code a bit.
Fixes#974.
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.
Since the new hwaccel API is now merged in ffmpeg's stable release, we can
finally remove support for the old API.
I pretty much kept lu_zero's new code unchanged and just added some error
printing (that we had with the old glue code) to make the life of our users
less miserable.
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 ALSA plugins take non-interleaved audio, but treat it as
interleaved, which results in various funny bugs. Users keep hitting
this issue, and it just doesn't seem worth the trouble.
CC: @mpv-player/stable
Prior to this commit we had a list of key modifiers and checked against that.
Actually, the Cocoa framework has a built in way to do it and it involves
calling performKeyEquivalent: on the menu instance.
Fixes#946
cc @mpv-player/stable: this should apply with no conflicts
This happens apparently randomly with rtmp:// and after seeks. This
eventually leads to audio decoding returning an EOF status, which
basically disables audio sync. This will lead to audio desync, even if
audio decoding later "recovers" when the demuxer actually returns audio
packets.
Hack-fix this by special-casing EAGAIN.
This didn't work, because the timebase was wrong. According to the
ffmpeg doxygen, if the stream index is -1 (which is what we used), the
timebase is AV_TIME_BASE. But this didn't work, and it really expected
the stream's timebase. Quite "surprising", since this feature
(avio_seek_time) is used by rtmp only.
Fixing this properly is too hard, so hack-fix our way around it.
STREAM_CTRL_SEEK_TO_TIME is also used by DVD/BD, so a new
STREAM_CTRL_AVSEEK is added. We simply pass-through the request
verbatim.
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.
With software decoding, images were uploaded to vdpau surfaces as they
were queued to the VO. This makes it slightly more complicated
(especially later on), and has no advantages - so stop doing it.
The only reason why this was done explicitly was due to attempts to keep
the code equivalent (instead of risking performance regressions). The
original code did this naturally for certain reasons, but now that we
can measure that it has no advantages and just requires extra code, we
can just drop it.
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.
Usually mp_image_params_guess_csp takes care of finding *some* default
for a video channel, but files with no video (or with extremely broken
configurations) end up leaving the colorspace information as
MP_CSP_PRIM_AUTO, which has no associated primaries.
As a result of this, color managed OSD messages could not display
because they were being color managed to match the (non-existing/absurd)
video channel. With this change, such non-spaces will have BT.709
primaries as far as color management and the OSD is concerned.
This fixes#961.
CC: @mpv-player/stable
Signed-off-by: wm4 <wm4@nowhere>
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.
Apparently this switch means all mouse input should be strictly
rejected. Some VO backends (such as X11) explicitly disable all mouse
events if this option is set, but others don't. So check them in
input.c, which increases consistency.
Follow up on commit 760548da. Mouse handling is a bit confusing, because
there are at least 3 coordinate systems associated with it, and it
should be cleaned up. But that is hard, so just apply a hack which gets
the currently-annoying issue (VO backends needing access to the VO) out
of the way.
Add an option that enables using native PulseAudio auto-updated timing
information, instead of the manual calculations added in mplayer2 times.
You can use --ao=pulse:no-latency-hacks to enable the new code. The code
is almost the same as the code that was removed with commit de435ed5,
but I didn't readd some bits I didn't understand. Likewise, the option
will disable the code added with that commit.
In my tests this seemed to work well, though the A/V sync display looks
funny when seeking.
The default is still the old behavior.
See issue #959.
This was needed by very old (0.9) versions only. Get rid of it.
Unfortunately, I can't cross-check with the original bug report, since
the bug URL leads to this:
Internal Server Error
TracError: IOError: [Errno 2] No such file or directory: '/home/lennart/svn/trac/pulseaudio/VERSION'
The windows message loop now runs in a separate thread. Rendering,
such as with Direct3D or OpenGL, still happens in the main thread.
In particular, this should prevent the video from freezing if the
window is dragged. (The reason was that the message dispatcher won't
return while the dragging is active, so mpv couldn't update the
video at all.)
This is pretty "rough" and just hacked in, and there's no API yet to
make this easier for other backends. It will be cleaned up later
once we're sure that it works, or when we know how exactly it should
work. One oddity is that OpenGL is actually completely managed in the
renderer thread, while e.g. Cocoa (which has its own threading code)
creates the context in the GUI thread, and then lets the renderer
thread access it.
One strange issue is that we now have to stop WM_CLOSE from actually
closing the window. Instead, we wait until the playloop handles the
close command, and requests the VO to shutdown. This is done mainly
because closing the window apparently destroys it, and then WM_USER
can't be handled anymore - which means the playloop has no way to
wakeup the GUI thread. It seems you can't really win here... maybe
there's a better way to have a thread receive messages with and
without a window, but I didn't find one yet.
Dragging the window (by clicking into the middle of it) behaves
strangely in wine, but didn't before the change. Reason unknown.
VO backends which are or will run in their own thread have a problem
with vo_mouse_movement() calling vo_control(). Restrict this to VOs
which actually need this.
win32 does not provide a proper per-window context pointer. Although it
does allow passing a user-chosen value to WM_CREATE/WM_NCCREATE, this
is not enough - the first message doesn't even have to be WM_NCCREATE.
This gets us in trouble later on, so go the easy route and just use a
TLS variable.
__thread is gcc specific, but Windows is a very "special" platform
anyway. We support only MinGW and Cygwin on it, so who cares. (C11
standardizes __thread as _Thread_local; we can use that later.)
Plan 9 has a very interesting synchronization mechanism, the
rendezvous() call. A good property of this is that you don't need to
explicitly initialize and destroy a barrier object, unlike as with e.g.
POSIX barriers (which are mandatory to begin with). Upon "meeting", they
can exchange a value.
This mechanism will be nice to synchronize certain stages of
initialization between threads in the following commit.
Unlike Plan 9 rendezvous(), this is not implemented with a hashtable,
because that would require additional effort (especially if you want to
make it actually scele). Unlike the Plan 9 variant, we use intptr_t
instead of void* as type for the value, because I expect that we will be
mostly passing a status code as value and not a pointer. Converting an
integer to void* requires two cast (because the integer needs to be
intptr_t), the other way around it's only one cast.
We don't particularly care about performance in this case either. It's
simply not important for our use-case. So a simple linked list is used
for waiters, and on wakeup, all waiters are temporarily woken up.
This shouldn't change anything. But it's worth making this explicit,
since it's very subtle and unintuitive: if the X parameter is the
magic value CW_USEDEFAULT, then the Y parameter is used as nCmdShow
parameter to ShowWindow(). And in our case, this is SW_HIDE (0),
because we want to create a hidden window.
This looked a bit overcomplicated. We don't care about the window
position (it should always be 0/0, unless the parent program moved it,
which it shouldn't). We don't care about the global on-screen position.
Also, we will just retrieve a WM_SIZE message if our window is resized,
and we don't need to update it manually.
The only thing we have to do is making sure our window fills the parent
window completely.
CS_OWNDC will make GetDC() always return the same HDC. This might
become a problem when OpenGL rendering and window management are
on different threads. Although I'm not too sure about this; our
code never requests a HDC outside of the OpenGL backend, and it
depends on whether win32 will internally request DCs. But in any
case, this seems to be cleaner.
Move the GL pixelformat setup code to gl_w32.c, where it's actually
needed. This also fixes that SetPixelFormat() should be called only
once, and not every time video params change.
These mostly describe self-explanatory things, and fail to explain
actually tricky things. Which means you just waste your time reading
this, and have to figure it out from the code anyway.
Preparation for moving win32 windowing to a separate thread.
The codesize is reduced a bit, because some small functions are
inlined, which reduces noise.
The main change is that now most functions use the private struct
directly, instead of accessing it indirectly through vo->w32.
Accesses to vo are minimalized.
The final goal is adding some sort of new windowing backend API. It
would be cleaner to use that as context pointer for all functions
(like struct vo was previously used), but since this is work in
progress, we just go with this commit.
ao_null is used to stop autoprobing (if all AOs before fail to init).
After it come things like ao_pcm, which should never be automatically
selected.
Remove a certain theoretically possible failure case, and force "some"
fallback.
mp_make_wakeup_pipe() always fails on win32. If this call fails on Linux
(and e.g. ao_alsa is used), this will probably burn CPU since poll()
won't work on the invalid file descriptor, but whatever, the failure
case is obscure enough.
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.
Accidentally broken in b6af44d3. For ad_lavc (and in general), the PTS
was not updated correctly when filtering only parts of audio frames,
and for ad_mpg123 and ad_spdif the PTS was additionally offset by the
frame size.
This could lead to incorrect time display, and possibly broken A/V sync.
Execute the format change based on whether we logically detected EOF
(after filters), instead of when the decode buffer was drained. It's
slightly cleaner. (The requirement of len>0 existed before.)
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.
Move a function call, which does not change semantics.
Write the extra buffer sample count in a more straight-forward way; the
old code was not meaningful in any way (anymore).
It's true that the decoder can successfully decode, but return no data
(for various reasons). We don't need to handle this specially, though.
We just let the decoder decode some more data. This doesn't increase the
danger of an endless loop either, because audio_decode() already calls
this function until enough is decoded.
"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.)
Useful for Windows stuff. Actually, ENCA support should catch this, but,
well, whatever, everyone seems to hate ENCA.
Detection with BOM is trivial, although it needs some hackery to
integrate it with the existing autodetection support. For one, change
the default value of --sub-codepage to make this easier.
Probably fixes issue #937 (the second part).
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.
The travis guys were so nice to activate multi OS support for us (it's a beta
feature). So now we build on OS X ass well to check for OS X specific breakage.
Later I might investigate further and build with the minimum supported SDK
version so that we don't break older systems by using newer Cocoa features.
This commit mainly moves the initial decoding of data (done to probe the
audio format) to generic code. This will make it easier to make audio
decoding non-blocking in a later commit.
This commit also changes how decoders return data: instead of having
them write the data into a prepared buffer, they return a reference to
an internal buffer (by setting dec_audio.decoded). This makes it
significantly easier to handle audio format changes, since the decoders
don't really need to care anymore.
If the decoder didn't set a samplerate, it was initialized from the
container samplerate.
This probably didn't make much sense, because it's passed to the
decoder on initialization (so it could definitely use it). It's an
artifact from commit 66a9eb57 (which removed some Matroska-specific non-
sense), and I've never seen it actually happen since it was made into a
warning. Just get rid of it.
This tells the demuxer thread that it should seek, instead of waiting
until the demuxer thread is ready.
Care has to be taken about the state between seek request and actual
seeking: newly demuxed packets have to be discarded. We can't just
flush when doing the actual seek, because the user thread could read
these packets.
I'm wondering if this could lead to issues due to relaxed ordering of
operations. But it should be fine, since seeking influences packet
reading only, and seeking is always strictly done before that.
Currently, this will have no advantages; unless audio is disabled. Then
seeking as well as normal playback can be non-blocking.
The MPlayer style syntax ("-mf fps=10:type=png") was removed a while
ago, and now only the flat variants ("--mf-fps=10" etc.) work.
CC: @mpv-player/stable
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.
Instead of starting to fill the packet queue if at least 1 stream is
selected, wait until there is at least 1 stream had new packets
requested.
In theory this is cleaner, because it allows you to e.g. do a seek and
then reselect streams without losing packets. Seeking marks all streams
as inactive, and without this new logic, the thread would read new
packets anyway right after seek.
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.
This fixes the same symptom as the previous commit, but when the demuxer
thread is enabled. In this case, if nothing was read from the demuxer,
the STREAM_CTRLs weren't updated either. To the player, this looked like
the stream cache was never making progress, so playback was kept paused.
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.
It can happen that read_packet() doesn't read a packet, even if it
succeeds. Typically this is because a packet was read, but then thrown
away, because it's not part of a selected stream. The result would be a
bogus EOF condition.
Fix by explicitly checking for EOF.
In corner cases, it might be possible that a demux_read_packet_async()
call fails to make the demuxer thread to read more packets.
If a packet is queued, the function will simply return a packet, without
marking the stream as active. As a consequence, read_packet() might
decide not to read any further packets, and the demuxer will never read
a packet and wake up the playback thread.
This was originally done to align it with demux_read_packet() semantics;
just drop this.
demux_read_any_packet() attempts to call read_packet(), but if no stream
is active, it can decide not to read anything. The function will return
NULL, which implies EOF. Fix this by explicitly
setting demux_stream->active if needed.
Also use dequeue_packet() instead of demux_read_packet(), because it's
cleaner. (Shouldn't change behavior.)
Possibly fixes#938.
We told the demuxer that a pipe (if stream cache is enabled) is
seekable. This is because the stream cache is technically seekable, it's
just that seeking may fail at runtime if a non-cached byte range is
requested.
This caused libavformat to issue seeks on initialization (at least when
piping mp4 youtube videos). Initialization failed completely after
spamming tons of error messages.
So, if an unseekable stream is cached, tell the demuxer that the file is
not seekable. This gets reversed later (when printing a message about
caching an unseekable stream), so the user can still try his luck by
issuing a seek command. The important part is that libavformat
initialization will not take code paths that will unnecessarily seek for
whatever reasons.
CC: @mpv-player/stable: regression from 0.3.x
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.
It was easy to get into a wakeup feedback loop on EOF. The reason that
EOF is complicated is that we try to retry reading when EOF is reached,
in case the EOF state actually disappears (e.g. when watching a
currently downloaded file).
This feature is probably worthless, since in practice you have to do a
seek to "unstuck" it anyway, but since the old code also did this, we
want to keep this behavior for now.
Avoid the feedback loop by introducing another EOF flag (last_eof), that
contains the actual previous EOF state, and is not overwritten when
retrying reading. Wakeup is skipped if the EOF state didn't change.
Also, actually call the wakeup callback when EOF is detected.
The line that adds "ds->active = false;" actually does nothing, but in
theory it's cleaner.
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.
The old FFmpeg API and the new Libav API disagree about mp4 display
rotation direction. Well, whatever, fix it trial-and-error-style.
CC: @mpv-player/stable: add
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).
For OGG audio files, we usually merge the per-stream metadata back to
the file-global metadata. Don't do that for OGM, because with OGM most
metadata is actually per-stream.
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
Logic for this was missing from pull.c. For push.c it was missing if the
driver didn't support it. But even if the driver supported it (such as
with ao_alsa), strange behavior was observed by users. See issue #933.
Always check explicitly whether the AO is in paused mode, and if so,
don't drain.
Possibly fixes#933.
CC: @mpv-player/stable
libdvdnav can actually jump into the middle of the DVD (e.g. scene
selection menus do that). Then time display is incorrect: we start from
0, even though playback time is somewhere else. This really matters when
seeking. If the display time mismatches, a small relative seek will
apparently jump to the beginning of the movie.
Fix this by initializing the PTS stuff on opening. We have to do this
after some small amount of data has been read from the stream (because
libdvdnav is crap and doesn't always update the time between seeks and
the first read; also see STREAM_CTRL_GET_CURRENT_TIME remarks in
cache.c; although this was not observed when testing with scene
selection menus). On the other hand, we want to do it before opening the
demuxer, because that will read large amounts of data and likely will
change the stream position.
Also see commit 49813670.
Happens when playing from a pipe.
Note that seeking forward doesn't work. It would be possible to create a
workaround for that by reading and skipping data until the target
position is reached (and writing the skipped data into the cache file),
but I'm not sure about that.
Fixes#928.
CC: @mpv-player/stable
There is no standard mechanism for detecting endianess. Doing it at
compile time in a portable way is probably hard. Doing it properly
with a configure check is probably hard too. Using the endian
definitions in <sys/types.h> (usually includes <endian.h>, which is
not available everywhere) works under circumstances, but the previous
commit broke it on OSX.
Ideally all code should be endian dependent, but that is not possible
due to the dependencies (such as FFmpeg, some video output APIs, some
audio output APIs).
Create a header osdep/endian.h, which contains various fallbacks.
Note that the last fallback uses libavutil; however, it's not clear
whether AV_HAVE_BIGENDIAN is a public symbol, or whether including
<libavutil/bswap.h> really makes it visible. And in fact we don't want
to pollute the namespace with libavutil definitions either. Thus it's
only the last fallback.
_GNU_SOURCE defines the kitchen sink, and also prefers glibc definitions
where glibc and POSIX conflict. Even though POSIX is worth less than
toilet paper, we still prefer the POSIX definitions.
rar.c needs asprintf(), which is _GNU_SOURCE-only. So we define
_GNU_SOURCE too specifically for this file.
This approach is similar to what other vo_opengl backends do. It can also be
used in the future to create another cocoa backend that renders offscreen
with IOSurfaces or FBOs.
Technically needed, but not strictly. It seems it works without in
practice, because demux_lavf.c reads exactly one packet for fill_buffer
call, so there are never packets queued.
We used a complicated and approximate method to cache the stream
timestamp, which is basically per-byte. (To reduce overhead, it was only
cached per 8KB-block, so it was approximate.)
Simplify this, and read/keep the timestamp only on discontinuities. This
is when demux_disc.c actually needs the timestamp.
Note that caching is currently disabled for dvdnav, but we still read
the timestamp only after some data is read. libdvdread behaves well, but
I don't know about libbluray, and the previous code also read the
timestamp only after reading data, so try to keep it safe.
Also drop the start_time offset. It wouldn't be correct anymore if used
with the cache, and the idea behind it wasn't very sane either (making
the player to offset the initial playback time to 0).
Key bindings are decided on the "down" event, so if the prefix is not
unique, the first/shortest will be used (e.g. when both "a" and "a-b"
are mapped, "a" will always be chosen).
This also breaks combining multiple mouse buttons. But it seems users
expect it to work, and it's indeed a bit strange that it shouldn't work,
as mouse bindings are emitted on the key "up" event, not "down" (if the
shorter binding didn't emit a command yet, why shouldn't it be
combinable).
Deal with this by clearing the key history when a command is actually
emitted, instead of when a command is decided. This means if both
MOUSE_BTN0 and MOUSE_BTN0-MOUSE_BTN1 are mapped, the sequence of holding
down BTN0 and then BTN1 will redecide the current command. On the other
hand, if BTN0 is released before BTN1 is pressed, the command is
emitted, and the key history is deleted. So the BTN1 press will not
trigger BTN0-BTN1.
For normal keys, nothing should change, because commands are emitted on
the "down" event already, so the key history is always cleared.
Might fix#902.
CC: @mpv-player/stable (if this fix is successful)
Doesn't work quite right, and will pause for the latency duration after
seeking. Some users use --ao=null to disable audio (even though they
should probably use --no-audio), and this use-case is broken by this
issue too.
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 work now, at least kind of. Note that actual success depends
on the behavior of the underlying lib{dvd{nav,read},bluray}
implementation, which could go very wrong.
In the worst case, it could happen that the underlying implementation
seeks a long time before the seek target time. In this case, the player
will just decode video until the target time is reached, even if that
requires e.g. decoding 30 mintues of video before refreshing.
In the not-so-bad but still bad case, it would just miss the seek
target, and seek past it.
In my tests, it works mostly ok, though. Seeking backwards usually
fails, unless something like --hr-seek-demuxer-offset=1 is used (this
makes it seek to 1 second before the target, which may or may not be
enough to compensate for the DVD/BD imprecision).
This is a pretty big change. Instead of doing a half-hearted passthrough
of the playback timestamp, we attempt to rewrite the raw MPEG timestamps
such that they match with the playback time.
We add the offset between raw start timestamp and playback time to the
packet timestamps. This is the easy part; but the problem is with
timestamp resets. We simply detect timestamp discontinuities by checking
whether they are more than 500ms apart (large enough for all video
faster than 2 FPS and audio with reasonable framesizes/samplerates), and
adjust the timestamp offset accordingly.
This should work pretty well. There may be some problems with subtitles.
If the first packet after a timestamp reset is a subtitle instead of
video, it will fail. Also, selecting multiple audio or video streams
won't work (but mpv doesn't allow selecting several anyway). Trying to
demux subtitles with no video stream enabled will probably fail.
Untested with Bluray, because I have no Bluray sample.
Background:
libdvdnav/libdvdread/libbluray make this relatively hard. They return a
raw MPEG (PS/TS) byte stream, and additionally to that provide a
function to retrieve the current "playback" time. The playback time is
what should be displayed to the user, while the MPEG timestamps can be
completely different. Even worse, the MPEG timestamps can reset. Since
we use the libavformat demuxer (instead of parsing the MPEG packets in
the DVD/BD code), it's hard to associate between these timestamps. As a
result, the time display is special cased in the playloop, and of low
quality (updates only all 1 or 2 seconds, sometimes is incorrect). The
fact that the stream cache can be between demuxer and the stream source
makes things worse.
All the libs seem to provide an event that tells whether timestamps are
resetting. But since this signalling is byte based, it's hard to connect
it to the demuxed MPEG packets. It might be possible to create some sort
of table mapping file positions to discontinuities and new timestamps.
(For simplicity, this table could be 2 entries large, sufficient to
catch all discontinuities if the distance between them is larger than
the total buffering.)
It can happen that demux_fill_buffer() adds more than 1 packet, and then
the packets would add up. Affects demux_disc.c only (nothing else uses
this function).
When seeking, we violently destroy the filter, because vapoursynth has
no proper API for terminating a video with unknown frame count. This
looks like an error to vapoursynth, and the error is returned via the
frame callbacks. The bug is that we remember this error state across
reinitialization, so on the first filter call after reinitialization, we
thought filtering the current frame failed. This caused a shift by 1
frame on each seek.
CC: @mpv-player/stable
Not all compilers on all platforms have atomics available (even if they
could, technically speaking).
We don't use atomics that much, only the following things rely on it:
1. the audio pull code, and all audio outputs using it
2. updating global msg levels
3. reading log messages through the client API
Just disable 1. and 3. if atomics are not available. For 2., using fake-
atomics isn't too bad; at worst, message levels won't properly update
under certain situations (but most likely, it will work just fine).
This means if atomics are not available, the client API function
mpv_request_log_messages() will do nothing.
CC: @mpv-player/stable
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.
Suggested by tholin on github issue #882.
This is not entirely clean, but the fields we're accessing might be
considered internal to libavformat. On the other hand, existence of the
fields is guaranteed by the ABI, and nothing in the libavformat doxygen
suggestes they're not allowed to be accessed.
CC: @mpv-player/stable
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.
(Again.)
This time, we simply make it event-based, as it should be. This is done
for both demuxer metadata and stream metadata.
For some ogg-over-icy streams, 2 updates are reported on stream start.
This is because libavformat reports an update right on start, while
including the same info in the "static" metadata. I don't know if that's
a bug or a feature.
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.
Recently, libavformat added demuxers to open image files like normal
demuxers. This is a good thing, but for now they interfere with the
operation of demux_mf. Add them to the blacklist until there is a proper
solution.
(The list doesn't contain _all_ recognized image formats, just those
that might interfere with demux_mf.)
CC: @mpv-player/stable
This returned a stream error value directly to libavformat, which can't
make sense. For example STREAM_ERROR (0) means success in libavformat
error codes. (The meaning of the libavformat read_seek return value is
underdocumented too.)
The intention is to make it obvious which mpv releases certain changes
will apply to.
Also attempt to fix RST formatting of the list. This is not very proper,
but probably good enough.
Cast away the "extra" bits (since apparently Window/XID is always
32 bit unsigned). This is not striclty needed, because you're not
supposed to pass garbage to --wid, just because the upper bits are
possibly not interpreted. But if you do so, this change increases
consistency in behavior and removes a strange behavior that was
thought to be a bug.
Also see github issue #906.
Don't use _x_arguments, as we don't support X arguments.
Get rid of -s, because we don't support multiple single-letter options
in one argument.
Add -S, because we ignore options after "--".
Completion now uses "--opt=value" instead of "--opt value". Once the
user presses space and starts a new argument, the option just
completed is out of the picture, whether or not it was given an
argument. This handles options with no arguments or optional arguments
much better; previously, completing such an option would effectively
disable completion for the next argument.
Custom completed options such as "--ao" and friends will no longer
claim to consume an extra argument.
Previous code was completly wrong. This still doesn't report the device
latency, but we report the buffer latency (as before the AO refactoring) and
the AudioUnit's latency (this is a new 'feature').
Apparently we can also report the device actual latency and we should also
calculate the actual sample rate of the audio device instead of using the
nominal sample rate, but I'll leave this for a later commit.
The mplayer1/2/mpv CoreAudio audio output historically contained both usage
of AUHAL APIs (these go through the CoreAudio audio server) and the Device
based APIs (used only for output of compressed formats in exclusive mode).
The latter is a very unwieldy and low level API and pretty much forces us to
write a lot of code for little workr. Also with the widespread of HDMI, the
actual need for outputting compressed audio directly to the device is getting
lower (it was very useful with S/PDIF for bandwidth constraints not allowing
a number if channels transmitted in LPCM).
Considering how invasive it is (uses hog/exclusive mode), the new AO
(`ao_coreaudio_device`) is not going to be autoprobed but the user will have
to select it.
When resizing the cache, the buffer for the DVD timestamps is
initialized with 0. This causes the player to always return playback
position 0 with any file format (not just DVD), and also makes all
relative seeks relative to position 0. Fix this by clearing the
timestamps explicitly.
Closes#899.
CC: @mpv-player/stable
Apparently clearing on every map can cause problems with vdpau when
switching virtual desktops and such. This was observed with at least
XMonad and nvidia-340.17. It's not observed on some other setups without
XMonad.
It's not clear why this happens. Normally, the window background is not
saved, so clearing should have no additional affect. It's a complete
mystery. Possible, the use of legacy X drawing commands (used to clear
the window) interferes with vdpau operation in non-trivial ways.
Work this around by clearing on initial map only. This probably only
hides the underlying issue, but good enough.
Closes#897.
CC: @mpv-player/stable
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.)
Some of these might be security relevant.
The RealAudio code was especially bad. I'm not sure if all RealAudio
stuff still plays correctly; I didn't have that many samples for
testing. Some checks might be unnecessary or overcomplicated compared
to the (obfuscated) nature of the code.
CC: @mpv-player/stable
rgain is not an additive value. It's a multiplier/gain.
Previous behaviour produced negative level values in some cases
(when rgain < 1.0) which caused volume to be louder when its value
was lowered.
CC: @mpv-player/stable
Signed-off-by: Mohammad Alsaleh <CE.Mohammad.AlSaleh@gmail.com>
Signed-off-by: wm4 <wm4@nowhere>
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).
Until now, the config functions added various allocations to the user-
provided talloc context. Make it so that they're all under the returned
allocation instead. This allows avoiding having to create an extra
temporary context for some callers, and also avoids adding random memory
leaks by accidentally passing a NULL context.
mp_find_all_config_files() has to be changed not to return a pointer
into the middle array for this to work. Make it add paths in order
(instead of reverse), and then reverse the array entries after that.
Also remove the declarations for the win-specific private functions.
Remove STRNULL(); it's barely needed anymore and the functions are
not called with NULL filenames anymore.
This means normally the XDG config dir will be used. But if the old
config dir (~/.mpv) exists and the XDG config dir does not, then don't
create it.
To simplify the code, also make mp_path_exists() accept NULL paths. In
that case it's considered as not existing. (Funnily, on Linux this
already worked, because the string is passed directly to the kernel,
and the kernel will just return EFAULT on invalid memory.)
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>
These consult the vertical resolution, matching against 576 for
PAL and 480/486 for NTSC. The documentation has also been updated.
Signed-off-by: wm4 <wm4@nowhere>
Notably, we now conform to SMPTE 428-1-2006 when decoding XYZ12 input,
and we can support rendering intents other than colorimetric when
converting between BT.709 and BT.2020, like with :srgb or :icc-profile.
With this change, XYZ input is directly converted to the output
colorspace wherever possible, and to the colorspace specified by the
tags and/or --primaries option, otherwise.
This commit also restructures some of the CMS code in gl_video.c to
hopefully make it clearer which decision is being done where and why.
This also avoids an extra matrix multiplication when using :srgb, making
that path both more efficient and also eliminating more hard-coded
values.
In addition, the previously hard-coded XYZ to RGB matrix will be
dynamically generated.
This add support for reading primary information from lavc, categorized
into BT.601-525, BT.601-625, BT.709 and BT.2020; and passes it on to the
vo. In vo_opengl, we always generate the 3dlut against the wider BT.2020
and transform our source into this colorspace in the shader.
For remarks, pretty much see the manpage additions. Could help with
network streams that require too much seeking (maybe), or might be
extended to help with the use case of watching and downloading a file
at the same time.
In general, it might be a useless feature and could be removed again.
It seems it's generally cleaner to leave this stuff to the BSDs.
Additionally, the NetBSD part wasn't even correct, because it made the
compiler prefer the system include path before local include files.
It is reasonably stable, so all further changes will be versioned.
Also change how the libmpv version number is generated. Fix the patch
version number to 0; I don't think we have a use for this. In
particular, the version doesn't version mpv, just the client API.
Do this simply by clearing the mapped buffer on every draw_image() call
without an actual video frame. (Maybe this is a bit expensive, but at
least not more expensive than regular video display.)
Clear the texture on reconfig(). (We could probably also do this simpler
with a flag, but this is actually less complicated - except that we move
the code to "map" a texture to a separate function.)
With the change to merge osd drawing into video frame drawing, some
bogus logic got in: they skipped drawing the OSD if no video frame is
available. This broke --no-video --force-window mode.
Until now, bitmap subtitles were decoded at "some" point, and then
simply replaced the old subtitle. Although the subtitle is selected
by time (PTS), it could happen that a subtitle was replaced too early.
One consequence is that this might lead to flicker even if the
subtitles are timed to follow each other without a gap (although most
subtitles are explicitly timed to introduce such a gap). With this
commit the past 4 subtitles are kept (instead of 1), so that the
correct one can be picked by time. This should fix the aforementioned
cases, but more importantly will allow demuxing/decoding and video
display to be somewhat asynchronous.
Still missing: somehow making sure the correct range of decoded
subtitles is available, instead of just passing along whatever comes
from the demuxer, and hoping that 4 queued subtitles are enough. But it
should certainly be good enough for now.
This removes a check that resets the subtitles if the PTS is 5 minutes
before the end of the current subtitle; this is probably not needed.
This fixes the build on platform where the atomic calls can't be turned into
lock-free instructions and thus need the external libatomic library (e.g. mips).
Make sure every video filter has valid parameters for input and output.
(This also ensures we don't take possibly invalid decoder output, or
feed invalid decodr/filter output to VOs.)
Also, the updated image size check now (almost) works like the
corresponding check in FFmpeg.
Until now, failure to allocate image data resulted in a crash (i.e.
abort() was called). This was intentional, because it's pretty silly to
degrade playback, and in almost all situations, the OOM will probably
kill you anyway. (And then there's the standard Linux overcommit
behavior, which also will kill you at some point.)
But I changed my opinion, so here we go. This change does not affect
_all_ memory allocations, just image data. Now in most failure cases,
the output will just be skipped. For video filters, this coincidentally
means that failure is treated as EOF (because the playback core assumes
EOF if nothing comes out of the video filter chain). In other
situations, output might be in some way degraded, like skipping frames,
not scaling OSD, and such.
Functions whose return values changed semantics:
mp_image_alloc
mp_image_new_copy
mp_image_new_ref
mp_image_make_writeable
mp_image_setrefp
mp_image_to_av_frame_and_unref
mp_image_from_av_frame
mp_image_new_external_ref
mp_image_new_custom_ref
mp_image_pool_make_writeable
mp_image_pool_get
mp_image_pool_new_copy
mp_vdpau_mixed_frame_create
vf_alloc_out_image
vf_make_out_image_writeable
glGetWindowScreenshot
The error log callback was not thread-safe and not library-safe. And
apparently there were some other details that made it not library-safe,
such as a global lcms plugin registry.
Switch the the thread-safe API provided by lcms2 starting with 2.6.
Remove our approximate thread-safety hacks.
Note that lcms basically provides 2 APIs now, the old functions, and
the thread-safe alternatives whose names end with THR. Some functions
don't change, because they already have a context of some sort. Care
must be taken not to accidentally use old APIs.
When the reader is out of data, it tries to wake up the cache thread to
get more data. In theory, there's a small race condition, which could
cause the cache to miss the wakeup and idle before reaction.
Most certainly didn't cause real issues, because even if this extremely
unlikely race condition happens, the cache won't idle for longer than
1 second (the hardcoded cache idle time).
OSD used to be not thread-safe at all, so a track was used to get it
redrawn. This mostly reverts commit 6a2a8880, because OSD not being
thread-safe was the non-trivial part of it.
Mostly untested, because this code path is used on OSX only, and I don't
have OSX.
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.
We certainly don't want to maintain and improve the internal converter,
but we still need the internal one for Libav. (In the Libav case,
demux_subreader.c will be used to read the MicroDVD file.)
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.
Subsurfaces are only used by the wayland vo. Thats why it makes sense to move
all osd and subsurface specific parts to the vo_wayland.c
Also destroy the subsurfaces and subcompositor properly.
Not all the hardware supports kCGLPFASupportsAutomaticGraphicsSwitching
(apparently all Mid-2010 and before MacBooks do not work with it), so fallback
to not asking for this attribute in the GL pixel format.
Well, not sure if this really improves anything, but at least it's less
of a WTF to the playback core than always returning the same timestamp
for every frame.
There's really no need to convert this to float and then back. This is
mostly of cosmetic nature, double precision was probably enough to avoid
rounding.
This affects packed RGB formats up to 16 bits per pixel. The old mplayer
names used LSB-to-MSB order, while FFmpeg (and some other libraries) use
MSB-to-LSB.
Nothing should change with this commit, i.e. no bit order or endian bugs
should be added or fixed. In some cases, the name stays the same, even
though the byte order changes, e.g. RGB8->BGR8 and BGR8->RGB8, and this
affects the user-visible names too; this might cause confusion.
These suffixes are annoying when they're redundant, so strip them
automatically. On little endian machines, always strip the "le" suffix,
and on big endian machines vice versa (although I don't think anyone
ever tried to run mpv on a big endian machine).
Since pixel format strings are returned by a certain function and we
can't just change static strings, use a trick to pass a stack buffer
transparently. But this also means the string can't be permanently
stored by the caller, so vf_dlopen.c has to be updated. There seems
to be no other case where this is done, though.
Not sure how this symbol becomes visible in glibc (probably accidental
or mandatory recursive inclusion via the other standard or Linux-
specific headers), but normally this include file is needed to get the
symbol.
The "classic" sub-option stuff is not really needed anymore. The only
remaining use can be emulated in a simpler way. But note that this
breaks the --screenshot option (instead of the "flat" options like
--screenshot-...). This was undocumented and discouraged, so it
shouldn't affect anyone.
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.
This means use of the min/max fields can be dropped for the flag option
type, which makes some things slightly easier. I'm also not sure if the
client API handled the case of flag not being 0 or 1 correctly, and this
change gets rid of this concern.
Also clarify the semantics.
It seems --idx didn't do anything. Possibly it used to change how the
now removed legacy demuxers like demux_avi used to behave. Or maybe
it was accidental.
--forceidx basically becomes --index=force. It's possible that new
index modes will be added in the future, so I'm keeping it
extensible, instead of e.g. creating --force-index.
Probably "needed" to get the correct alignment, although I'm not aware
of actual breakages or performance issues.
In fact we should probably always just allocate AVPackets, but for now
use the simple fix.
FFmpeg requires a bullshit padding after each input buffer, and they
just increased that padding without warning and without ABI or API bump.
We need this only in one file (although mp_image hardcodes something
similar, for which no FFmpeg API define is available), so drop our own
define.
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).
Unfortunately, there's a recursive function call in ao_lavc.c (play
function), leading to a deadlock. The locking is getting a bit messy, so
just make the lock recursive.
This fixes#844.
Squashed from the following mplayer-svn commits. The '#' is removed from
the bug ID to prevent github from doing something stupid. Instead of
adding the mplayer configure check for clock_gettime(), the POSIX
identifiers are used for checking presence of the function.
Use correct type of timestamps when recording from v4l2. Fix 2176
Patch by Jarek Czekalski <jarekczek at poczta onet pl>.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@37222 b3059339-0415-0410-9bf9-f77b7e298cf2
Allow building of v4l2 without clock_gettime().
Add overly verbose message in case monotone timestamps are required by the kernel.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@37223 b3059339-0415-0410-9bf9-f77b7e298cf2
Similar to previous commits.
This also renames --doubleclick-time to --input-doubleclick-time, and
--key-fifo-size to --input-key-fifo-size. We could keep the old names,
but these options are very obscure, and renaming them seems better for
consistency.
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.
Additionally to removing the global variables, this makes the options
more uniform. --ssf-... becomes --sws-..., and --sws becomes --sws-
scaler. For --sws-scaler, use choices instead of magic integer values.
Usually, each stream driver declares the size and option list of its
private data. This was pretty natural for when most streams still used
global variables to setup their defaults. They did by pointing
priv_defaults to the (mutable) struct containing the option values. But
falls short when storing the option values in MPOpts. So provide a
somewhat inelegant but simple way to let the stream implementation setup
the priv struct at initialization time.
This is done with the get_defaults callback. It should return a copy of
the struct used in MPOpts. (A copy, because if MPOpts is changed, string
fields might be deallocated, and if that field is not described by
stream_info.options, it won't be copied on init.)
Pretty much nothing changes, but using -tv-scan with suboptions doesn't
work anymore (instead of "-tv-scan x" it's "-tv scan-x" now). Flat
options ("-tv-scan-x") stay compatible.
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.
The Perl script generating the completions actually invokes mpv, and it
runs during the build. This is not sane and breaks at least cross
compilation.
As a workaround, disable the completions by default for now.
Playing a video and then an audio file with cover art kept displaying
the last frame of the video. This was because the hasframe flag was set,
perhaps due to redrawing the last video frame before the cover art image
is decoded.
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.
--sub-file is actually a string list, so you can add multipel external
subtitle files. But to be able to set a list, the option value was split
on ",". This made it impossible to add filenames.
One possible solution would be adding escaping. That's probably a good
idea (and some other options already do this), but it's also complicated
both to implement and for the user.
The simpler solution is making --sub-file appending, and make it take
only a single entry.
I'm not quite sure about this yet. It breaks the invariant that if a
value is printed and parsed, you get the same value back. So for now,
just go with the simple solution.
Fixes#840.
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.
It wasn't necessarily clear how this works. Especially make clear that
the API user shouldn't expect that there's one byte per readable event
in the wakeup pipe. Actually, the reason why this currently won't work
is because property notifications can generate more events than wakeups.
The limit of 4096 is a more fundamental issue, but the event ringbuffer
is currently limited to 1000 entries anyway.
Also add some important comments to mpv_set_wakeup_callback.
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.
This was never intended to be installed; waf just picked it up
automagically. There's also a closed ticket on github where someone
complains that the program "simple" is installed, and I didn't realize
at this point that it was actually installed by default when enabling
the client API.
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.)
It seems we can't really get rid of this. There are no other hints to
remove decorations that work across all reasonable WMs, so we're stuck
with the ugly motif stuff.
But at least we can make the code for it less ugly.
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.
(The old "force" choice of that option is renamed to "force-default".)
This allows overriding native ASS script subtitle styles with the style
provided by the --sub-text-* options (like --sub-text-font etc.). This
is disabled by default, and needs to be explicitly enabled with the
--ass-style-override=force option and input property.
This uses in fact exactly the same options (--sub-text-*) and semantics
as the ones used to configure unstyled text subtitles.
It's recommended to combine this with this in the mpv config file:
ass-force-style="ScaledBorderAndShadow=1" # work around dumb libass behavior
Also, adding a key binding to toggle this behavior should be added,
because overriding can easily break:
L cycle ass-style-override
This would cycle override behavior on Shift+L and allows quickly
disabling/enabling style overrides.
Note: ASS should be considered a vector format rather than a subtitle
format. There is no easy or reliable way to determine whether the style
of a given subtitle event can be changed without destroying visuals or
not. This patch relies on a simple heuristic, which often works and
often breaks.
This could for example happen when serving an incomplete file from http,
and the demuxer tries reading data from the end of the file when opening
it (e.g. with avi). Seeking past EOF fails with http, so the file could
never be opened, and the cache would get stuck trying to seek to the
position.
We can't really make the cache report seek failure directly (it would
suck for various reasons), so just make the cache report EOF if seeking
fails.
The previous commit assumed the filter would be 1x1 (then constant
weight is correct) - but our code in fact uses at least a 2x2 filter. A
1x1 filter would generally be useless, except for nearest scaling - so
it didn't exist.
Insteasd of adding such a 1x1 filter, just turn the nearest weight
function into a scare function, which should take care of the issue.
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.
This accidentally rejected d==0. We can actually deal with sub-normals
fine, we just want to exclude nan and infinity (although infinity is
already accounted for, but anyway).
If a single person complains, I will readd it. But I don't expect that
this will happen.
The main reason for removing this is that it's some of the most unclean
code remaining, it's unmaintained, and I've never ever heard of someone
using it.
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 would imply eglGetProcAddress() doesn't work correctly, but using
dlsym() does. For now get rid of it - it won't work in libmpv, and we'll
probably need a better workaround if it's still broken.
This code was in the initial wayland commit.
The functions glXGetProcAddressARB() and glXQueryExtensionsString() were
loaded using dlsym(). This could fail when compiling to libmpv, because
then dlopen(NULL, ...) will look in the main program's list of
libraries, and the libGL linked to libmpv is never considered. (Don't
know if this somehow could be worked around.) The result is that using
vo_opengl with libmpv can fail.
Avoid this by not using dlsym(). glXGetProcAddressARB() was already used
directly in the same file, and that never caused any problems. (Still
add it to the configure test.) glXQueryExtensionsString() is documented
as added in GLX 1.1 - that's ancient.
So the device buffer can be refilled quickly. Fixes dropouts in certain
cases: if all data is moved from the soft buffer to the audio device
buffer, the waiting code thinks it has to enter the mode in which it
waits for new data from the decoder. This doesn't work, because the
get_space() logic tries to keep the total buffer size down. get_space()
will return 0 (or a very low value) because the device buffer is full,
and the decoder can't refill the soft buffer. But this means if the AO
buffer runs out, the device buffer can't be refilled from the soft
buffer. I guess this mess happened because the code is trying to deal
with both AOs with proper event handling, and AOs with arbitrary
behavior.
Unfortunately this increases latency, as the total buffered audio
becomes larger. There are other ways to fix this again, but not today.
Fixes#818.
Apparently this can happen. So actually only return from waiting if ALSA
excplicitly signals that new output is available, or if we are woken up
externally.
This call was used limited the buffer size if installed RAM was below 16
MB. This stopped being useful a decade ago. The check could also
overflow on 32 bit systems. Just get rid of it.
This did not flush remaining audio in the buffer correctly (in case an
AO has an internal block size). So we have to make the audio feed thread
to write the remaining audio, and wait until it's done.
Checking the avoid_ao_wait variable should be enough to be sure that all
data that can be written was written to the AO driver.
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.
Until now, we've always calculated a timeout based on a heuristic when
to refill the audio buffers. Allow AOs to do it completely event-based
by providing wait and wakeup callbacks.
This also shuffles around the heuristic used for other AOs, and there is
a minor possibility that behavior slightly changes in real-world cases.
But in general it should be much more robust now.
ao_pulse.c now makes use of event-based waiting. It already did before,
but the code for time-based waiting was also involved. This commit also
removes one awkward artifact of the PulseAudio API out of the generic
code: the callback asking for more data can be reentrant, and thus
requires a separate lock for waiting (or a recursive mutex).
uint_least32_t could be larger than uint32_t, so the return values of
mp_ring_get_wpos/rpos must be adjusted. Actually just use unsigned long
as type instead, because that is less awkward than uint_least32_t.
This was originally added because we thought this would make a good
portable audio API, which would give us good behavior on Windows, Linux,
and OSX. But this hope was disappointed: it's not reliable enough (nice
deadlocks on Linux when seeking, i.e. resetting the audio device),
doesn't have enough features (no channel maps, no digital passthrough),
and in general just is not very good.
There were subtle and minor race conditions in the pull.c code, and AOs
using it (jack, portaudio, sdl, wasapi). Attempt to remove these.
There was at least a race condition in the ao_reset() implementation:
mp_ring_reset() was called concurrently to the audio callback. While the
ringbuffer uses atomics to allow concurrent access, the reset function
wasn't concurrency-safe (and can't easily be made to).
Fix this by stopping the audio callback before doing a reset. After
that, we can do anything without needing synchronization. The callback
is resumed when resuming playback at a later point.
Don't call driver->pause, and make driver->resume and driver->reset
start/stop the audio callback. In the initial state, the audio callback
must be disabled.
JackAudio of course is different. Maybe there is no way to suspend the
audio callback without "disconnecting" it (what jack_deactivate() would
do), so I'm not trying my luck, and implemented a really bad hack doing
active waiting until we get the audio callback into a state where it
won't interfere. Once the callback goes from AO_STATE_WAIT to NONE, we
can be sure that the callback doesn't access the ringbuffer or anything
else anymore. Since both sched_yield() and pthread_yield() apparently
are not always available, use mp_sleep_us(1) to avoid burning CPU during
active waiting.
The ao_jack.c change also removes a race condition: apparently we didn't
initialize _all_ ao fields before starting the audio callback.
In ao_wasapi.c, I'm not sure whether reset really waits for the audio
callback to return. Kovensky says it's not guaranteed, so disable the
reset callback - for now the behavior of ao_wasapi.c is like with
ao_jack.c, and active waiting is used to deal with the audio callback.
We don't need to combine __sync_add_and_fetch with a memory barrier,
since these intrinsics are documented as using a full barrier already.
Use __sync_fetch_and_add instead of __sync_add_and_fetch; this gives
atomic_fetch_add() the correct return value (although we don't use it).
Use __sync_fetch_and_add to emulate atomic_load(). This should enforce
the full barrier semantics better. (This trick is stolen from the
FreeBSD based stdatomic.h emulation.)
In most places where af_fmt2bits is called to get the bits/sample, the
result is immediately converted to bytes/sample. Avoid this by getting
bytes/sample directly by introducing af_fmt2bps.
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.
This is incomplete; the video chain will still hold some vaapi objects
after destroying the decoder and thus the vaapi context. This is very
bad. Fixing it would require something like refcounting the vaapi
context, but I don't really want to.
mpv supports two hardware decoding APIs on Linux: vdpau and vaapi. Each
of these has emulation wrappers. The wrappers are usually slower and
have fewer features than their native opposites. In particular the libva
vdpau driver is practically unmaintained.
Check the vendor string and print a warning if emulation is detected.
Checking vendor strings is a very stupid thing to do, but I find the
thought of people using an emulated API for no reason worse.
Also, make --hwdec=auto never use an API that is detected as emulated.
This doesn't work quite right yet, because once one API is loaded,
vo_opengl doesn't unload it, so no hardware decoding will be used if the
first probed API (usually vdpau) is rejected. But good enough.
So, basically this worked only with streams that were not local files,
because stream_dvd.c "intercepts" local files to check whether they
point to DVD images. This means if a stream is not writeable, we have to
try the next stream implementation.
Unbreaks 2-pass encoding.
Sometimes, Matroska files store monotonic PTS for h264 tracks with
b-frames, which means the decoder actually returns non-monotonic PTS.
Handle this with an evil trick: if DTS is missing, set it to the PTS.
Then the existing logic, which deals with falling back to DTS if PTS is
broken. Actually, this trick is not so evil at all, because usually, PTS
has no errors, and DTS is either always set, or always unset. So this
_should_ provoke no regressions (famous last words).
libavformat actually does something similar: it derives DTS from PTS in
ways unknown to me. The result is very broken, but it causes the DTS
fallback to become active, and thus happens to work.
Also, prevent the heuristic from being active if PTS is merely monotonic
instead of strictly-monotonic. Non-unique PTS is broken, but we can't
fallback to DTS anyway in these cases.
The specific mkv file that is fixed with this commit had the following
fields set:
Muxing application: libebml v1.3.0 + libmatroska v1.4.1
Writing application: mkvmerge v6.7.0 ('Back to the Ground') [...]
But I know that this should also fix playback of mencoder produced mkv
files.
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'
We pass a pointer to a GLint to sscanf, using the %d format. That format
_always_ takes int, and not GLint (whatever the heck that is). If GLint
is always int, then it doesn't make a difference, but is still better
because it doesn't play russian roulette with pointers.
Don't emit "hard" references to OpenGL functions. Always use the
platform specific function to lookup OpenGL functions, such as
glXGetProcAddress() with GLX (x11).
This actually fixes the build if only Wayland is enabled (e.g. using
--disable-gl-x11 on Linux).
Note that some sources claim that wglGetProcAddress() (win32) does not
return function pointers for OpenGL 1.1 functions (even if they are
valid and necessary in OpenGL 3.0). But if that happens, the fallback
employed in gl_w32.c/w32gpa() should catch this.
Setting this property was added 12 years ago, and the code was always
incorrect. The underlying data type is "long", not "pid_t". It's well
possible that the data types are different, and the pointer to the pid
variable is directly passed to XChangeProperty, possibly invoking
undefined behavior.
It's funny, because in theory using pid_t for PIDs sounds more correct.
Binding multiple commands at once where always considered not
repeatable, because the MP_CMD_COMMAND_LIST wasn't considered
repeatable.
Fixes#807 (probably).
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.
This will load other files in the same directory when a single file is
played. It's an often requested feature, but we definitely don't want it
in the core.
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.
A playlist_move command that moves an entry onto itself (both arguments
have the same index) should do nothing, but it did something broken. The
underlying reason is that it checks the prev pointer of the entry which
is temporarily removed for moving.
Also sneak in some cosmetics.
setmode() exists on Windows/msvcrt only, so there's no need for a
config test.
I couldn't reproduce the problem with seekable pipes on wine, so axe
it. (I'm aware that it still could be an issue on real Windows.)
For some reason, we support writeable streams. (Only encoding uses that,
and the use of it looks messy enough that I want to replace it with FILE
or avio today.)
It's a chaos: most streams do not actually check the mode parameter like
they should. Simplify it, and let streams signal availability of write
mode by setting a flag in the stream info struct.
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).
This is mostly covered by the OSX workaround, if the timeout is very
high. It also means that with systems using 32 bit time_t, the time will
overflow 2036 already, instead of 2037, but we don't consider this a
problem.
_WIN_LAYER is apparently an old GNOME thing (also explains why there is
a function vo_x11_get_gnome_layer() involved in this code). Prefer the
NetWM hints over this. This just moves the NetWM case if-body over the
_WIN_LAYER one.
You can't use identifiers starting with "_" and an uppercase letter in
application programs. They are reserved by the C standard.
Unrelated change: drop unused/misleading vo_wm_NETWM define.
I can only assume the old code was wrong. EWMH does not document
anything with _WIN_LAYER. Instead, you have to toggle the state using a
client message. We also remove these weird non-sense fallbacks, like
using _NET_WM_STATE_BELOW - what the hell?
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
Some operating systems apparently can't deal with really long timeouts
in pthread_cond_timedwait(). Passing a time about 300000 in the future
makes the call return immediately. (tv_sec/time_t doesn't overflow in
this situation.) Reduce the wait time to about 100 days, which seems
to work fine.
The list of affected OSes follows: OSX
Probably more correct and better readable. Although the special-casing
of 0x1p63 is weird in terms of readability (the value itself is
INT64_MAX+1, so it's already outside of range, but INT64_MAX is not
exactly representable with double precision).
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.
This usually can't happen, because even if time_us (first input value)
is INT64_MAX, the value added to tv_sec will be about 2^43, and tv_sec
will be <2^31, far below a possible overflow in 64 bits. But should
time_t be 32 bits (32 bit Linux/Windows?), an overflow could happen.
Integrate it with the existing surface allocator in vdpau.c. The changes
are a bit violent, because the vdpau API is so non-orthogonal: compared
to video surfaces, output surfaces use a different ID type, different
format types, and different API functions.
Also, introduce IMGFMT_VDPAU_OUTPUT for VdpOutputSurfaces wrapped in
mp_image, rather than hacking it. This is a bit cleaner.
black_pixel is an (apparently necessary) 1x1 black surface used for
clearing the screen. It was allocated in RGB mode only, but is sometimes
used in YUV mode too.
In at least 2 places, we want to "shift" positional parameters by a
certain amount, or remove all parameters if there are fewer parameters.
bash handled this just fine, but dash vomits upon it. POSIX shell allows
both behaviors, so we have to fix it.
Fix a comparison done with "==" instead of "=".
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.
Without this change, the compiler uses by default the "talloc.h" file
installed by the package libtalloc within /usr/local/include. Found and
tested on OpenBSD but FreeBSD has the same patch on its ports tree.
This should be more correct. The builtins were made to directly map to
C11, and the way we use them is now relatively close to how gcc
implements atomics in 4.9. In particular, we make use of the load and
store builtins.
I'm not entirely sure why gcc didn't support stdatomic.h in 4.8 already.
Maybe support for the builtins was incomplete or broken - so there's a
lot of room for doubt about the correctness of this.
In my opinion, we shouldn't use atomics at all, but ok.
This switches the mpv code to use C11 stdatomic.h, and for compilers
that don't support stdatomic.h yet, we emulate the subset used by mpv
using the builtins commonly provided by gcc and clang.
This supersedes an earlier similar attempt by Kovensky. That attempt
unfortunately relied on a big copypasted freebsd header (which also
depended on much more highly compiler-specific functionality, defined
reserved symbols, etc.), so it had to be NIH'ed.
Some issues:
- C11 says default initialization of atomics "produces a valid state",
but it's not sure whether the stored value is really 0. But we rely on
this.
- I'm pretty sure our use of the __atomic... builtins is/was incorrect.
We don't use atomic load/store intrinsics, and access stuff directly.
- Our wrapper actually does stricter typechecking than the stdatomic.h
implementation by gcc 4.9. We make the atomic types incompatible with
normal types by wrapping them into structs. (The FreeBSD wrapper does
the same.)
- I couldn't test on MinGW.
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 works around an issue in OpenBox: OpenBox apparently sizes the
normal window incorrectly if aspect ratio hints are set, and the window
size is off by 1 pixel. Then, when going fullscreen and leaving
fullscreen again, mpv sets the hints based on OpenBox' broken window
size, and as result, OpenBox sizes the window incorrectly and is off by
1 pixel again - so it's 2 pixels off in total. The error gets more
visible, the more often you toggle fullscreen mode.
Work this around by not setting the window hints if we don't need to.
Actually we only need to do this when the video is resized during
fullscreen, which happens rarely. Under normal circumstances, leaving
fullscreen mode requires that the WM restores the old state.
As such, this commit is not only a workaround, but actually a cleanup.
Note that we do need to set the hints when leaving fullscreen if the
window has resized: even though we set the hints in
vo_x11_highlevel_resize (called by vo_x11_config_vo_window), this
doesn't seem to have an effect (at least on IceWM), so we have to do it
after that.
Side note: ot seems commit 625ad57a strangely triggered the OpenBox
issue according to user reports; I'm not sure why.
Drop: sami, vplayer, rt, pjs, mpsub, aqt, jacosub. None of these seem
to be actually in use, except sami. Sami is very complex, and the
results subreader produces are not very useful.
For all these formats, there are still parsers in FFmpeg. We remove the
subreader implementation, because it might contain security relevant
bugs and such. (This is old, unmaintained C string parsing code, written
in times where absolutely nobody cared about security. The kind of
awesome code.)
We keep the other formats, because they're (mostly) commonly used and
relatively simple, for UTF16 support (still missing in FFmpeg), and for
the sake of Libav.
Before this commit, this was somehow polled (i.e. not the right way).
Also, selects the correct window when doing --wid=0 (which is another
weird special-case).
Enabling DPMS even though you disabled it globally is pretty unfriendly,
so don't do it. Instead, we only disable DPMS if it was enabled, and
only enable it if we disabled it ourselves.
The other way should never happen (disabling DPMS permanently), unless
mpv crashes during playback.
It can easily happen that mp_time_us_to_timespec() gets a time in the
past, and then the time difference will be negative. Regression
introduced in commit f47a4fc3.
Also fix an underflow check in mp_add_timeout().
Currently, it's unused. Still keep the file, because it's not unlikely
we'll need it again, and removing/readding the include statements for
this file is too annoying.
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.
Reduces some code-duplication.
Just call DPMSEnable/DPMSDisable, instead of DPMSForceLevel when
reenabling DPMS. "Force" sounds evil, and messing with DPMS is already
pretty evil. I'm not even sure that we should.
XGetWindowProperty is a really bad API, almost as if the NSA designed
it. The wrapper takes care of verifying the return values and handle
corner cases.
The window "gravity" influences how placement interacts with WM added
borders (i.e. from decorations). This is probably what the code removed
in commit c14721c8 was about.
In theory, we'd probably want to set the gravity depending on the
relative placement requested by the user (so that it's possible to line
up the top/left video pixel with the monitor corner, as well as the
bottom/right pixel - but that would be too complicated, and who cares
after all?).
I'm also not sure whether CenterGravity really uses the top/left corner
as reference point (instead of making coordinates relative to the window
center), but empirically it's correct.
Try to get the "new" code path (using NetWM/EWMH) free of hacks done for
the sake of old WMs or the no-WM case.
Implement --fs-screen using _NET_WM_FULLSCREEN_MONITORS.
Keeps the window centered on resize. Seems nicer. (Although it's worse
if 1. the default placement of the WM puts it into a monitor corner,
and 2. you switch to a larger video.)
It was added with 3813c685 in 2004. I'm not really sure why this gravity
stuff would be needed; apparently it has to do with misplacements with
broken WMs and had to be changed on fullscreen. Just get rid of it; it
works perfectly fine without on modern WMs.
The thread discussing this is here:
http://mplayerhq.hu/pipermail/mplayer-dev-eng/2004-July/027674.html
When writing a video to foo.mp3, the user's intention is clearly to drop
the video stream, and similarly, when writing to foo-%d.png, the
intention is clearly to drop the audio stream. Now, explicit
specification of --no-audio or --no-video is no longer necessary in
these cases.
This should get rid of some flickering. Since this actually skips all
the wacky fullscreening code on startup, this might lead to certain
wacky features to stop working. In this case, you'll have to use the
--x11-fstype option, and disable _NETWM_STATE_FULLSCREEN usage.
vo_x11_map_window() was attempting to clear the window on map. However,
it did so immediately after the map request. It probably assumed that
the drawing calls for clearing the window would be queued along with the
map request, and then executed in the right order. However, this
assumption was wrong - the map request first has to go to the window
manager (I guess?), so a lot of things happen before the window is even
mapped.
Fix this by moving the call to the MapNotify message handler, when the
window (apparently) becomes really visible.
I also tried to set CWBackPixel to black instead, but this seemed to
result in flickering on manual resizing.
This blocks everything, until the window is actually reported as mapped.
This fixes the race condition between VO initialization and mapping the
window, which resulted in possibly different window sizes, leading to an
immediate redraw, visible as flashing.
Note that if the map event never comes for some reason, we're out of
luck and will block forever.
It could in theory happen that the filter loop will enter a blocking
wait, even though it could make progress by emptying the list of
already-filtered images. I'm not quite sure if this could actually cause
a real issue - probably not.
VapourSynth won't just filter multiple frames at once on its own. You
have to request multiple frames at once manually. This is what this
commit introduces: a sub-option controls how many frames will be
requested at once. This also changes the semantics of the maxbuffer sub-
option, now renamed to buffered-frames.
The situation has changed a bit since the days of mplayer2, so we can
use more/less diplomatic wording. Merge the two sections listing
changes from MPlayer and mplayer2. Mention the client API and Lua
scripting as alternatives to slave mode.
I'm calling MPlayer code "horrible". This is not meant as an offense,
but after turning around almost every line of MPlayer code, I believe
I have a right to say this. Sorry. I would say that MPlayer has a
surprisingly sane and simple architecture (for what it is), but much
of it drowned under a load of evil hacks or not-cleaned-up-yet code.
This started as a bunch of smaller changes to make the old configure
script maintainable with minimum effort. It ended up as complete
rewrite, because at once point I started to like shell programming (I
hope this sickness is curable), and I wanted to see how small I can
make the configure script. The typical configure test is now 1 or 2
lines big, located in 1 or 2 places, instead of >15 lines and being
spread over 5 or 6 places.
The main "trick" is factoring the tests into a few generic, commonly
needed tests, instead of writing everything manually.
A lot of effort was spent on making the waf based build-system work
properly on all supported platforms, while the old configure script was
neglected. It seems that nobody maintains the non-Linux parts of the
configure script anymore, and all improvements go into the waf scripts.
Thus it makes no sense anymore to maintain the non-Linux parts. They're
just dead weight. Remove them completely.
Also apply some additional simplifications. For example, listing
enabled/disabled VO modules seems like a waste of effort.
These were in the old configure script too.
Two flags are explicitly tested, because I have no idea how widespread
support for them is, and testing them is just easier than trying to look
them up in various gcc/clang manuals. There are people using gcc 4.2
out there, so some caution is warranted.
waf has a proper test, but no test was added to old-configure.
This means that on OSX, old-configure is not supposed to work anymore.
But still allow it to compile cleanly on OSX.
Previous to this commit, read_chunk was not set in stream_smb. The
cache was therefore filled in small 8K chunks. This resulted in poor
performance when compared to, for example, smbnetfs on the same
network.
The value of 128k is chosen both because it is emperically
the "levelling off point" for throughput into mpv's cache, and because
it is the value chosen by smbnetfs when serving smb shares to
mpv.
Note that this change has no effect unless --cache is explicitly
specified as smb:// streams do not activate cache by default. This is
because the default cache size of 320K is so small it actually makes
smb:// perfomance worse. For best results use at least --cache=1024.
This didn't quite work. The main issue was that get_space tries to be
clever to reduce overall buffering, so it will cause the playloop to
decode and queue only as much audio as is needed to refill the AO in
reasonable time. Also, even if ignoring the problem, the logic of the
previous commit was slightly broken. (This required a few retries,
because I couldn't reproduce the issue on my own machine.)
When the audio buffer went low, but could not be refilled yet, it could
happen that the AO playback thread and the decode thread could enter a
wakeup feedback loop, causing up to 100% CPU usage doing nothing. This
happened because the decoder thread would wake up the AO thread when
writing 0 bytes of newly decoded data, and the AO thread in reaction
wakes up the decoder thread after writing 0 bytes to the AO buffer.
Fix this by waking up the decoder thread only if data was actually
played or queued. (This will still cause some redundant wakeups, but
will eventually settle down, reducing CPU usage close to ideal.)
This shouldn't matter, but it's probably better if the code to check is
valid - otherwise an extremely clever compiler might fail to compile it,
and the feature would be misdetected. (Probably.)
Found by cppcheck.
Caused failure to detect .pls files, because they were misdetected as
m3u. The problem is that "forcing playlist files" and "forcing a
specific playlist format" are not really treated separate, and in both
cases p->force is set to true. This made m3u detect all files as m3u
if --playlist was used. So correctly check whether the file format is
actually being probed or not.
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.
I don't think this is really a very good idea because it is conceptually
incorrect but other prominent multimedia programs use this approach
(VLC and xbmc), and it seems to make the conversion more robust in certain
cases.
For example it has been reported, that configuring a receiver that can output
7.1 to output 5.1, will make CoreAudio report 8 channel descriptions, and the
last 2 descriptions will be tagged kAudioChannelLabel_Unknown.
Fixes#737
The code was falling back to the full waveext chmap_sel when less than 2
channels were detected. This new code is slightly more correct since it only
fills the chmap_sel with the stereo or mono chmap in the fallback case.
CoreAudio supports 3 kinds of layouts: bitmap based, tag based, and speaker
description based (using either channel labels or positional data).
Previously we tried to convert everything to bitmap based channel layouts,
but it turns out description based ones are the most generic and there are
built-in CoreAudio APIs to perform the conversion in this direction.
Moreover description based layouts support waveext extensions (like SDL and
SDR), and are easier to map to mp_chmaps.
This might shift bits into the sign, which is undefined behavior. Making
the right operand unsigned was supposed to help with this, but it seems
it did nothing, and C99 makes the result type dependent on the left
operand only.
Use the exact floating point value, instead of a broken integer
constant. The expression calculating the constant probably relied on
undefined behavior, because it left-shifts a negative value.
This also changes the type of the constant to double, which is perfectly
fine, and maybe better than an integer constant.
Preparation so that various things related to video can run in different
threads. One part to this is making the video surface pool safe.
Another issue is the preemption mechanism, which continues to give us
endless pain. In theory, it's probably impossible to handle preemption
100% correctly and race-condition free, unless _every_ API user in the
same process uses a central, shared mutex to protect every vdpau API
call. Otherwise, it could happen that one thread recovering from
preemption allocates a vdpau object, and then another thread (which
hasn't recovered yet) happens to free the object for some reason. This
is because objects are referenced by integer IDs, and vdpau will reuse
IDs invalidated by preemption after preemption.
Since this is unreasonable, we're as lazy as possible when it comes to
handling preemption. We don't do any locking around the mp_vdpau_ctx
fields that are normally immutable, and only can change when recovering
from preemption. In practice, this will work, because it doesn't matter
whether not-yet-recovered components use the old or new vdpau function
pointers or device ID. Code calls mp_vdpau_handle_preemption() anyway to
check for the preemption event and possibly to recover, and that
function acquires the lock protecting the preemption state.
Another possible source of potential grandiose fuckup is the fact that
the vdpau library is in fact only a tiny wrapper, and the real driver
lives in a shared object dlopen()ed by the wrapper. The wrapper also
calls dlclose() on the loaded shared object in some situations. One
possible danger is that failing to recreate a vdpau device could trigger
a dlclose() call, and that glibc might unload it. Currently, glibc
implements full unloading of shared objects on the last dlclose() call,
and if that happens, calls to function pointers pointing into the shared
object would obviously crash. Fortunately, it seems the existing vdpau
wrapper won't trigger this case and never unloads the driver once it's
successfully loaded.
To make it short, vdpau preemption opens up endless depths of WTFs.
Another issue is that any participating thread might do the preemption
recovery (whichever comes first). This is easier to implement. The
implication is that we need threadsafe xlib. We just hope and pray that
this will actually work. This also means that once vdpau code is
actually involved in a multithreaded scenario, we have to add
XInitThreads() to the X11 code.
Use the newly provided mp_vdpau_handle_preemption() function, instead of
accessing mp_vdpau_ctx fields directly. Will probably make multithreaded
access to the vdpau context easier.
Mostly unrelated to the actual changes, I've noticed that using hw
decoding with vo_opengl sometimes leads to segfaults inside of nvidia's
libGL when doing the following:
1. use hw decoding + vo_opengl
2. switch to console (will preempt on nvidia systems)
3. switch back to X (mpv will recover, switches to sw decoding)
4. enable hw decoding again
5. exit mpv
Then it segfaults when mpv finally calls exit(). I'll just blame nvidia,
although it seems likely that something in the gl_hwdec_vdpau.c
preemption handling triggers corner cases in nvidia's code.
This was broken for some time, and it didn't recover correctly.
Redo decoder display preemption. Instead of trying to reinitialize the
hw decoder, simply fallback to software decoding. I consider display
preemption a bug in the vdpau API, so being able to _somehow_ recover
playback is good enough.
The approach taking here will probably also make it easier to handle
multithreading.
Move the code that copies the dylib's to the bundle to a new script
(dylib-unhell.py) which is called by osxbundle.py.
dylib-unhell is about 20x faster than the previous implementation. This is
accomplished by removing superflous shell-out operations which are kept track
of using an in memory tree of all the needed dependencies. Moreover the
shell-outs have been further optimized by not requiring a complete shell for
every operation and just using subprocess.call (which is equivalent to Popen).
The many "boxes" in this entry were causing rst2pdf to fail because it
couldn't figure out where to break the page. Make the boxes smaller by
removing semi-redundant examples. Also try and make surrounding text a
little shorter by rewording.
This is done after filters, so things like framerate-doubling
deinterlacing is accounted for.
Unfortunately, framedropping can cause inaccuracies (especially after
precise seeks), and we can't really know when that happens. Even though
we know that the decoder might drop a frame if we request it to do so,
we don't know when the dropped frame will start or stop affecting the
video filter chain. Video filters can have frames buffered, and we
can't tell at which point the dropped frame would have been output.
It's not even possible to mark a discontinuity after seek, because
again we don't know if the filter chain still has the discontinuity
within its buffers.
So we have to live with the fact that the output of this property can
be completely broken after seek, unless --no-hr-seek-framedrop is used.
This allows disabling of decoder framedrop during hr-seek.
It's basically another useless option, but it will help exploring
whether this framedropping really makes seeking faster, or whether
disabling it helps with precise seeking (especially frame backstepping).
Until recently, the VO was an unavoidable part of the seeking code path.
This was because vdpau deinterlacing could double the framerate, and hr-
seek and framestepping etc. all had to "see" the additional frames. But
we've removed the frame doubling from the vdpau VO and moved it into a
video filter (vf_vdpaupp), and there's no reason left why the VO should
participate in seeking.
Instead of queuing frames to the VO during seek and skipping them
afterwards, drop the frames early.
This actually might make seeking with vo_vdpau and software decoding
faster, although I haven't measured it.
Now we avoid calling update_video() twice on reconfig (once to check
whether there are still new frames, and again to actually do the
reconfig). Instead, we check whether there's still something going on
before calling update_video() at all, and depending on that
update_video() will be allowed to reconfig or not.
This will simplify some things later.
It doesn't look like vo_wayland_config() necessarily sets this flag, so
it seems safer to trigger an explicit resize. This accounts for the case
when playing a new file with different size than the one before.
Currently, vo_reconfig() calculates the requested window size and sets
the vo->dwidth/dheight fields _if_ VOCTRL_UPDATE_SCREENINFO is
implemented by the VO or the windowing backend. The window size can be
different from the display size if e.g. the --geometry option is used.
It will also set the vo->dx/dy fields and read vo->xinerama_x/y.
It turned out that this is very backwards and actually requires the
windowing backends to workaround these things. There's also
MPOpts.screenwidth/screenheight, which used to map to actual options,
but is now used only to communicate the screen size to the vo.c code
calculating the window size and position.
Change this by making the window geometry calculations available as
separate functions. This commit doesn't change any VO code yet, and just
emulates the old way using the new functions. VO code will remove its
usage of VOCTRL_UPDATE_SCREENINFO and use the new functions directly.
Because the http playlist URL I had for testing claimed to be m3u by
file extension and mime type, but didn't have the header.
Note that this actually changes behavior only in the case the format is
detected by mime type. Then p->force will be set before calling the
parser, and the header becomes optional.
Changing --softvol-max and then resuming would change the volume level
on resume to something different than the original volume. This is
because the user volume setting is always between 0-100, and 100
corresponds to --softvol-max gain.
Avoid that changing -softvol-max and resuming an older file could lead
to a too loud volume level by refusing to restore if --softvol-max
changed.
Commit 433161 actually broke vo_opengl (and maybe others), because
config_ok is not necessarily set correctly yet _during_ reconfig. So a
vo_get_src_dst_rects() call during reconfig did nothing.
When the VO was not initialized with vo_reconfig(), or if the last
vo_reconfig() failed, changing panscan would cause a crash due to
vo_get_src_dst_rects() dereferencing vo->params (NULL if not
configured).
Just do nothing if that happens, as there is no video that could be
displayed anyway.
Doesn't really seem to be much of use. Get rid of the remaining uses of
it.
Concerning vo_opengl_old, it seems uninitGl() works fine even if called
before initialization.
Make it more suitable for chaining. This means a function formatting a
value to a string using a static buffer can work exactly like
mp_snprintf_append itself.
Also rename it to mp_snprintf_cat, because that's shorter.
Basically, extract the option table from DOCS/man/en/changes.rst, and
search the table if an option wasn't found. If there's an entry about
it, print it. Hopefully this behavior is slightly more userfriendly.
This is strictly bound to option names. It doesn't work for option
values, nor does it attempt to emulate the old option.
The comment says that it wakes up the main thread if 50% has been
played, but in reality the value was 0.74/2 => 37.5%. Correct this. This
probably changes little, because it's a very fuzzy heuristic in the
first place.
Also move down the min_wait calculation to where it's actually used.
mp3 has a hack lowering the probescore for format detection. This is
because detecting mp3s is hard due to their nature, and the fact that
ID3v2 tags are sometimes several megabytes big.
When playing mp3 from network, the mime-type is usually set, and that
matches the format hack entry meant for webradios, overriding the normal
mp3 entry. This can lead to network mp3s not being detected. Lower the
network case to the same probescore as on-disk mp3s. The difference is
that for network mp3s, we don't load the full probe-buffer, and we lower
the amount of audio the demuxer will read to collect data on opening
(0.5 seconds instead of typically 5 seconds).
This was a minor code duplication between vf_vdpaupp.c and vo_vdpau.c.
(In theory, we could always require using vf_vdpaupp with vo_vdpau, but
I think it's better if vo_vdpau can work standalone.)
Also remove MSGL_SMODE and friends.
Note: The indent in options.rst was added to work around a bug in
ReportLab that causes the PDF manual build to fail.
When loading a video, and a script reacts to MPV_EVENT_VIDEO_RECONFIG,
and the script inserts a video filter, the first frame can be skipped.
This happens simply because the first frame is (usually) still queued in
the video filter chain, and changing the filter chain will drop all
queued frames. So this is just a corner case that just happens in a
weird situation.
But it's still annoying when having such a script, and starting
something where the first frame is very visible, and not starting in
paused mode. (All in all, a corner case.) Do this by immediately queuing
1 filtered frame to the VO immediately after reconfig, instead of
leaving it to the video loop doing it as "incremental" work. Simply
fallthrough to the next case. We must not overwrite "r" in this case,
because that contains the current status.
Note that the first frame will not be filtered using the inserted
filter.
This wasn't really fine, and could (perhaps) cause weird corner cases on
reinit or when the player was paused.
Before eb9d20, video_left was also set to true if vo->frame_loaded was
set, and this variable basically indicated whether the previous
update_video() call was successful. This was overlooked when changing
everything. Simply always call update_video(), it should be equivalent.
This can happen when the input stream is somehow blocking on network,
and the user still send input in one way or another, and one of the
commands is a compound command ("cmd a ; cmd b").
Apparently the value of a pointer is "indeterminate" after a free()
call, even if you never dereference the pointer after the free. Since
talloc_free() calls free(), this applies here.
This affects the return value of mp.script_name, the "client name"
(what's returned by mpv_client_name()) and all associated features, as
well as the mpv terminal output module prefix when scripts print
something.
As discussed in #748.
Change how the video decoding loop works. The structure should now be a
bit easier to follow. The interactions on format changes are (probably)
simpler. This also aligns the decoding loop with future planned changes,
such as moving various things to separate threads.
vf_fix_img_params() takes care of overwriting image parameters that are
normally not set correctly by filters. But this makes no sense for input
images. So instead, check that the input is correct.
It still has to be done for the first input image, because that's used
to handle some overrides (see video_reconfig_filters()).
These replace vf_read_output_frame(), although we still emulate that
function. This change is preparation for another commit (and this is
basically just to reduce the diff and signal/noise ratio in that
commit).
Basically, if we feed the filter a new image even after the EOF state
has been reached (e.g. because the input stream "recovered"), we want
the filter to restart, instead of returning an error forever.
Some non-deinterlacing filters (potentially denoising) also use
additional frames for filtering. The vdpau docs suggest providing at
least 1 future and 2 past _fields_, which means we need to provide 1
past frame (the future field is already the other field of the current
field, and both fields are in the same frame).
We can easily achieve this by buffering an additional frame in the non-
deint case.
Remove the special casing of vo_vdpau vs. other VOs. Replace the
complicated interaction between vo.c and vo_vdpau.c with a simple queue
in vo.c. VOs other than vdpau are handled by setting the length of the
queue to 1 (this is essentially what waiting_mpi was).
Note that vo_vdpau.c seems to have buffered only 1 or 2 frames into the
future, while the remaining 3 or 4 frames were past frames. So the new
code buffers 2 frames (vo_vdpau.c requests this queue length by setting
vo->max_video_queue to 2). It should probably be investigated why
vo_vdpau.c kept so many past frames.
The field vo->redrawing is removed. I'm not really sure what that would
be needed for; it seems pointless.
Future directions include making the interface between playloop and VO
simpler, as well as making rendering a frame a single operation, as
opposed to the weird 3-step sequence of rendering, drawing OSD, and
flipping.
Give up on the deint_filters[] array, and probe using explicit code
instead. Add additional checks to test the pixel format to avoid
annoying warnings when a hardware deinterlacer is inserted when the
current video chain is obviously incompatible.
The previous commits changed vo_vdpau so that these options are set by
vf_vdpaupp, and the corresponding vo_vdpau were ignored. But for
compatibility, keep the "old" options working.
The value of this is questionable - maybe the vo_vdpau options should
just be removed. For now, at least demonstrate that it's possible.
The "deint" suboption still doesn't work, because the framerate doubling
logic required for some deint modes was moved to vf_vdpaupp. This
requires more elaborate workarounds.
This is slightly incomplete: the mixer options, such as sharpen and
especially deinterlacing, are ignored. This also breaks automatic
enabling of interlacing with 'D' or --deinterlace. These issues will be
fixed later in the following commits.
Note that we keep all the custom vdpau queue stuff. This will also be
simplified later.
This uses mp_vdpau_mixer_render(). The benefit is that it makes vdpau
deinterlacing just work. One additional minor advantage is that the
video mixer creation code is factored out (although that is a double-
edged sword).
This factors out some code from vo_vdpau.c, especially deinterlacing
handling. The intention is to use this for vo_vdpau.c to make the logic
significantly easier, and to use it for vo_opengl (gl_hwdec_vdpau.c) to
allow selecting deinterlace and postprocessing modes.
As of this commit, the filter actually does nothing, since both vo_vdpau
and vo_opengl treat the generated images as normal vdpau images. This
will change in the following commits.
It now inserts no filters and does nothing until the hot-key is pressed.
This makes it more suitable to be put in ~/.mpv/lua.
When the hot-key is pressed, it now inserts the cropdetect filter and
waits 1 second (or a --lua-opts specified duration) before gathering
the cropdetect metadata and inserting the appropriate crop filter. A
second press of the hotkey removes the crop.
It might have been nice not to do this so that metadata could
accumulate accross seeks, but it seems libavfilter looses its copy
anyway on recreate_graph.
lavfi would segfault due to a NULL dereference if it was asked for its
metadata and none had been allocated (oops). This happens for libav
which has no concept of filter metadata.
Commit 5e4e248 added a mp_image_params field to mp_image, and moved many
parameters to that struct. display_w/h was left redundant with
mp_image_params.d_w/d_h. These fields were supposed to be always in
sync, but it seems some code forgot to do this correctly, such as
vf_fix_img_params() or mp_image_copy_attributes(). This led to the
problem in github issue #756, because display_w/_h could become
incorrect.
It turns out that most code didn't use the old fields anyway. Just
remove them. Note that mp_image_params.d_w/d_h are supposed to be always
valid, so the additional checks for 0 shouldn't be needed. Remove these
checks as well.
Fixes#756.
In theory, returning the screenshot with original pixel aspect would
allow avoiding scaling them with image formats that support non-square
pixels, but in practice this isn't used anyway (nothing seems to
understand e.g. jpeg aspect ratio tags).
We only support them for input. The frame properties of output frames
are ignored (except frame durations).
Properties not set for now: _ChromaLocation, _Field, _FieldBased
Set _DurationNum/_DurationDen on each VS frame, instead of
_AbsoluteTime. The duration is the difference between the timestamp of
the frame and the next frame, and when receiving filtered VS frames, we
convert them back to an absolute PTS by summing them.
We pass the timestamps with microsecond resolution. mpv uses double for
timestamps internally, so we don't know the "real" timebase or FPS. VS
on the other hand uses fractions for frame durations. We can't pass
through the numbers exactly, but microseconds ought to be enough to be
even safe from accumulating rounding errors.
Since this leaks video images, and the player keeps feeding new images
to the fitler even if it fails, this would probably have disastrous
consequences.
Or in other words, add support for properly draining remaining frames
from video filters. vf_yadif is buffering at least one frame, and the
buffered frame was not retrieved on EOF.
For most filters, ignore this for now, and just adjust them to the
changed semantics of filter_ext. But for vf_lavfi (used by vf_yadif),
real support is implemented. libavfilter handles this simply by passing
a NULL frame to av_buffersrc_add_frame(), so we just have to make
mp_to_av() handle NULL arguments.
In load_next_vo_frame(), we first try to output a frame buffered in the
VO, then the filter, and then (if EOF is reached and there's still no
new frame) the VO again, with draining enabled. I guess this was
implemented slightly incorrectly before, because the filter chain still
could have had remaining output frames.
This extracts the scheduling logic to a single function which is nicer to keep
it consistent.
Additionally make sure we don't schedule sync operations from a sync operation
itself since that could cause deadlocks (even if it should not be happening
with the current code).
Previously the window could be made to completly exit the screen with a
combination or moving it close to an edge and halving it's size (via cmd+0).
This commit address the problem in the most simple way possibile by
constraining the window to the closest edge in these edge cases.
This fixes a couple of issues with the Cocoa `--native-fs` mode, primarily:
- A ghost titlebar at the top of the screen in full screen:
This was caused by the window constraining code kicking in during
fullscreen. Simply returning the unconstrained rect from the constraining
method fixes the problem.
- Incorrect behavior when using the titlebar buttons to enter/exit
fullscreen, as opposed to the OSD button.
This was caused by mpv's internal fullscreen state going out of sync with
the NSWindow's one. This was the case because `toggleFullScreen:`
completely bypassed the normal event flow that mpv expects.
Signed-off-by: Ryan Goulden <percontation@gmail.com>
Change style for mpv, simplify and refactor some of the constraining code.
Signed-off-by: Stefano Pigozzi <stefano.pigozzi@gmail.com>
I thought the "_" in "_AbsoluteTime" was part of the documentation
markup.
This still doesn't help us with VS filters that change timing;
apparently you must use frame durations instead.
Make the filter apply the pixel aspect ratio of the input to the output.
This is more useful than forcing 1:1 PAR when playing anamorphic video
such as DVDs.
VapourSynth itself actually allows passing through the aspect ratio, but
it's in a not very useful form for us: it's per video-frame instead of
constant (i.e in VSVideoInfo). As long as we don't have a way to allow a
filter to spontaneously change output parameters, we can't use this.
(And I don't really feel like making this possible.)
Our code currently tries to link -lpthread and adds stuff like -D_REENTRANT
based on the target platform.
GCC actually supports to just pass a -pthread compiler and linker flag that
will automatically enable threading and define the correct symbols for the
platform, so let's try to just use that as our first choice.
clang also supports -pthread but it must be used only as a compiler flag,
so we also take care of that scenario with this commit.
-Wempty-body is not available on all gcc versions but we were using it
unconditionally. Also remove the usage from the clang case. clang still
defines `__GNUC__` so it still gets all the gcc specific flags.
This should fix the build on systems with older gcc versions like OpenBSD which
still comes bundled with gcc 4.2 for license issues.
One problem is that for example stdio functions won't restart syscalls
manually, and instead treat EINTR as an error. So passing SA_RESTART is
the only sane thing to do, unless you have special requirements, which
we don't.
VP9 packets can contain 2 frames in some video packets (from which 1
frame is invisible). Due to a design mismatch between libvpx and the
libavcodec vp9 decoder, libvpx can take the "full" packets, but lavc vp9
can not. The consequence is that we have to split the packets if we want
to feed them to the lavc codec.
This is not entirely correct yet: timestamp handling is missing.
--demuxer=lavf and ffmpeg native utilities have the same problem. We can
fix this only once the ffmpeg VP9 parser is fixed.
This essentially reverts commit cca13efb. The code in the if was
supposed to be run only if the mouse button was down, because in this
case the mouse area is never considered to be left. Since it was run
for every mouse button, mouse_leave wasn't sent.
Fixes#745.
pthread_equal() returns 0 if the threads are not the same, and somehow
I got that wrong. The worst is that I actually explicitly checked the
manpage when I wrote this code.
This used an unnamed union, which is allowed in GNU C and C11, but not
C99. This broke the build with some older compilers.
Replaces pull request #744.
This reverts commit 6e34b0ec1f.
There has always been an error message "proxy already has a listener" and
I couldn't reproduce where it is comming from until now. The display interface
already has a listener and we can't overwrite it. Now remove the code and avoid
this error message.
Conflicts:
video/out/wayland_common.c
The interrupt callback will can be called from another thread if the
cache is enabled, and the stream disconnects. Then stream_reconnect()
will call this function from within the cache thread.
mp_input_check_interrupt() is not thread-safe due to read_events() not
being thread-safe. It will call input callbacks added with
mp_input_add_fd() - these callbacks lead to code not protected by locks,
such as reading X11 events.
Solve this by adding a stupid hack, which checks whether the calling
thread is the main playback thread (i.e. calling the input callbacks
will be safe). We can remove this hack later, but it requires at least
moving the VO to its own thread first.
This used global variables for the asynchronous interrupt callback.
Pick the simple and dumb solution and stuff the callback into
mpv_global. Do this because interrupt checking should also work in the
connect phase, and currently stream creation equates connecting.
Ideally, this would be passed to the stream on creation instead, or
connecting would be separated from creation. But since I don't know yet
which is better, and since moving stream/demuxer into their own thread
is something that will happen later, go with the mpv_global solution.
This is the only function which actually used the time argument of
stream_check_interrupt(). Considering that the whole player freezes
anyway, this is not worth the complication.
Also generally reduce the maximum wait time due to timeout. Introduce
exponential backoff, which makes the first reconnect retries faster, but
still waits up to 500ms in the later retries.
For some reason, some files appear to have broken mp3 packets, or at
least in a form that libavcodec can't deal with. The audio in the sample
file in question could not be decoded using libavcodec.
The problematic file had variable packet sizes, and the libavcodec
decoder kept printing "mp3: Header missing" for each packet it was fed.
Remuxing with mkvmerge fixes the problem. The mp3 data is probably not
VBR, and remuxing resulted in fixed-size mp3 frames. So I don't know why
the sample file was muxed this way - it might just be incorrect.
The sample file had "libmkv 0.6.4" as MuxingApp (although I could not
get mkvinfo to print this element, maybe the file uses an incorrect
element ID), and "HandBrake 0.9.4" as WritingApp.
Note that the libmpg123 decoder does not have any issues with it. It's
probably more robust, because libmpg123 was made to decode whole mp3
files, not just single frames.
Fixes issue #742.
The here documented guarantee might simplify code using this mechanism
a lot, because it becomes unnecessary to invent a separate mechanism to
make the mp_dispatch_queue_process loop exit after processing a dispatch
callback. (Instead, the dispatch callback can set a flag, and the caller
of mp_dispatch_queue_process can check it.)
This allows client API users and Lua scripts to side-step the pretty
horrible video filter string "language" (although it's back and can't be
avoided when using libavfilter).
The wakeup is needed to make mp_dispatch_queue_process() return if
suspension is not needed anymore - which is only the case when the
request count reaches 0.
The assertion added with this commit always has/had to be true.
Note that this mechanism is similarly "unreliable" as for example
pthread_cond_timedwait(). Trying to target the exact wait time will just
make it more complex.
The main use case for this is for threads which either use the dispatch
centrally and want mp_dispatch_queue_process to do a blocking wait for
new work, or threads which have to implement timers. For the former,
anything is fine, as long as they don't have to do active waiting for
new works. For the former, callers are better off recalculating their
deadline after every call.
This is much simpler, leaves fairness isues etc. to the operating
system, and will work better with threading-related debugging tools.
The "trick" to this is that the lock can be acquired and held only while
the queue is in suspend mode. This way we don't need to make sure the
lock is held outside of mp_dispatch_queue_process, which would be quite
messy to get right, because it would have to be in locked state by
default.
mp_dispatch_queue_process() releases the queue->lock mutex while
processing a dispatch callback. But this allowed mp_dispatch_lock() to
grab the "logical" lock represented by queue->locked. Grabbing the
logical lock is not a problem in itself, but it can't be allowed to
happen while the callback is still running.
Fix this by claiming the logical lock while the dispatch callback is
processed. Also make sure that the thread calling mp_dispatch_lock() is
woken up properly.
Fortunately, this didn't matter, because the locking function is unused.
This was part of osdep/threads.c out of laziness. But it doesn't contain
anything OS dependent. Note that the rest of threads.c actually isn't
all that OS dependent either (just some minor ifdeffery to work around
the lack of clock_gettime() on OSX).
Add the event FD after preinit, remove it before destroy. There's no
need to do it on vo_config, and there's no need to remove the event
FD when vo_config fails.
This should fix some issues, such as not being able to set the
"no-video" option with MPV_FORMAT_FLAG.
Note that this changes semantics a bit. Now setting an option strictly
overwrite it, even if the corresponding command line option does not.
For example, if we change --sub to append by default, then setting the
"sub" option via the client API would still never append. (Oddly, this
also applies to --vf-add, which will overwrite the old value when using
the client API.)
I'm doing this because there's no proper separation between the command
line parser and setting an option using the MPV_FORMAT_STRING format.
Maybe the solution to this mess would be adding format aware code (i.e.
m_option_set_node) to every option type, and falling back to strings
only if needed - but this would mean that you couldn't set e.g. an
integer option using MPV_FORMAT_STRING, which doesn't seem to be ideal
either.
In conclusion, the current approach seems to be most robust, but I'm
open to suggestions should someone find that these semantics are a
problem.
glob() is mandated by POSIX. For the only non-POSIX platform we support,
Windows, we have our own replacement. So the ifdeffery is not needed.
Still leave the checks in the configure scripts, because they have to
decide whether to compile the replacement or not. (Although this could
be special cased to mingw-only, the wscript seems to make this hard.)
Unfortunately, if a VO can't display something as intended, we can just
complain to the user, and leave it at it. But it's still better than
silently displaying things differently with different VOs.
For now, this is used for rotation only. Other things that we should
check includes colorspace and colorlevels stuff.
When using rotation with hw decoding, and the VO does not support
rotation, vf_rotate is attempted to be inserted. This will go wrong, and
after that it can't recover because a vf_scale filter was autoinserted.
Just removing all autoinserted filters before reconfig fixes this.
The Windows port uses CommandLineToArgvW, which doesn't expand wildcards
in command line arguments. Use glob to expand them instead, but only for
non-option arguments.
glob-win.c wasn't big, so it was easier to rewrite it. The new version
supports Unicode, handles directories properly, sorts the output and
puts all its allocations in the same talloc context to simplify the
implementation of globfree.
Notably, the old glob had error checking code, but didn't do anything
with the errors since the error reporting code was commented out. The
new glob doesn't copy this behaviour. It just treats errors as if there
were no more matching files, which shouldn't matter for mpv, since it
ignores glob errors too.
To match the other Windows I/O helper functions, the definition is moved
to osdep/io.h.
If the VO can't do rotation, insert a filter to do this. Note that this
doesn't reuse the filter insertion code from command.c (used by "vf"
input command), because that would end up more complicated: we don't
even want to change the user filter option.
This turned out much more complicated than I thought. It's not just a
matter of adjusting the texture coordinates, but you also have to
consider separated scaling and panscan clipping, which make everything
complicated.
This actually still doesn't clip 100% correctly, but the bug is only
visible when rotating (or flipping with --vf=flip), and using something
like --video-pan-x/y at the same time.
For rotation, we assume that the source image will be rotated within the
VO, so the aspect/panscan code needs to calculate its param using
rotated coordinates. VOs which support rotation natively can use this.
This couldn't rotate by 180°. Add this, and also make the parameter in
degrees, instead of magic numbers.
For now, drop the flipping stuff. You can still flip with --vf=flip or
--vf=mirror. Drop the landscape/portrait stuff - I think this is
something almost nobody will use. If it turns out that we need some of
these things, they can be readded later.
Make it use libavfilter. Its vf_transpose implementation looks pretty
simple, except that it uses slice threading and should be much faster.
Often, user configs set options that are not suitable for encoding.
Usually, playback and encoding are pretty different things, so it makes
sense to keep them strictly separate. There are several possible
solutions. The approach taken by this commit is to basically ignore the
default config settings, and switch to an [encoding] config profile
section instead. This also makes it impossible to have --o in a config
file, because --o enables encode mode.
See github issue #727 for discussion.
Fix all include statements of the form:
#include "libav.../..."
These come from MPlayer times, when FFmpeg was somehow part of the
MPlayer build tree, and this form was needed to prefer the local files
over system FFmpeg.
In some cases, the include statement wasn't needed or could be replaced
with mpv defined symbols.
Not needed anymore. I'm not opposed to having asm, but inline asm is too
much of a pain, and it was planned long ago to eventually get rid fo all
inline asm uses.
For the note, the inline asm use that was removed with the previous
commits was almost worthless. It was confined to video filters, and most
video filtering is now done with libavfilter. Some mpv filters (like
vf_pullup) actually redirect to libavfilter if possible.
If asm is added in the future, it should happen in the form of external
files.
No change in speed (or even slightly faster, though I tested with
progressive solid color video only), and normally we use libavformat's
vf_pullup anyway.
I didn't test the speed, but by default, this filter diverts to
libavfilter already. So this would help only if libavfilter is disabled,
or libavfilter doesn't have vf_noise (like on Libav). For these cases,
we still provide the (possibly but not necessarily) slower C
implementation of vf_noise.
This makes it multiple times slower. However, the output format (packed
YUV) isn't handled efficiently by anything to begin with, and I have no
clue we even have this filter. I guess it's one of these filters which
find some use sometimes, but are not of higher importance, which
justifies removing the faster inline asm.
This can be used to easily extent the status line for one's own needs.
I'm not experienced with lua so a few things could probably be done a
better way.
This is for the sake of multi-key commands again. This could break:
SPACE ignore
SPACE-SPACE command
while this worked:
SPACE-SPACE command
SPACE ignore
The reason being that if the shorter command was first in the list,
it would obviously match, and searching was stopped.
This is for the sake of multi-key combinations (see github issue #718).
Now a multi-key sequence isn't matched if any of the previous keys were
actually mapped.
We only need to track key up/down for a single key. There may be some
minor loss of robustness, but this can (probably) happen only if a VO or
user sends strange or complicated sequences of events. In the normal
case, what we do is more than enough. Most simplification comes from the
fact that mpv is not a game console, and users normally execute only one
action at once.
The window close button is usually mapped to the CLOSE_WIN pseudo-key.
Until now, --input-test treated this pseudo-key like any other key (like
the rest of the input handling code), so you couldn't close the window
in this mode. The manpage had silly instructions and warnings how to
deal with this.
Just always quit when CLOSE_WIN is received, and improve the
instructions.
The input code always supported combinations of multiple keys (even in
MPlayer, although there the code was active really only for mouse
buttons). This was arcance and also made the code more complicated. I
only know of a single person who ever made use of this feature.
Remove this feature, and repurpose some of the support code (e.g.
parsing, display of key combinations, etc.) to handle such multi-
combinations as sequences, instead of keys to be pressed at the same
time. This is much simpler and implements the feature requested in
github issue #718.
This commit will probably cause a bunch of regressions, since the input
handling code has some weird corner cases. I couldn't find any problems
when testing, though.
This avoids too many realloc() calls if the caller is appending to an
audo buffer. This case is actually quite noticeable when using something
that buffers a large amount of audio.
For some reason, the buffered_audio variable was used to "cache" the
ao_get_delay() result. But I can't really see any reason why this should
be done, and it just seems to complicate everything.
One reason might be that the value should be checked only if the AO
buffers have been recently filled (as otherwise the delay could go low
and trigger an accidental EOF condition), but this didn't work anyway,
since buffered_audio is set from ao_get_delay() anyway at a later point
if it was unset. And in both cases, the value is used _after_ filling
the audio buffers anyway.
Simplify it. Also, move the audio EOF condition to a separate function.
(Note that ao_eof_reached() probably could/should whether the last
ao_play() call had AOPLAY_FINAL_CHUNK set to avoid accidental EOF on
underflows, but for now let's keep the code equivalent.)
This replaces translate_key_input with a solution that gives mpv more
control over how keyboard input is converted to unicode. As a result:
- Key up/down events are generated the same way for all keys.
- Dead keys generate their base character instead of being combined with
the following character.
- Many Ctrl and Ctrl+Alt key combinations that were previously broken
are fixed, since it's possible to discover the base keys.
- AltGr doesn't produce special characters when mp_input_use_alt_gr is
false.
This also fixes some logic to do with detecting AltGr and adds proper
UTF-16 decoding.
This was reported with PulseAudio 2.1. Apparently it still has problems
with reporting the correct delay. Since ao_pulse.c still has our custom
get_delay implementation, there's a possibility that this is our fault,
but this seems unlikely, because it's full of workarounds for issues
like this. It's also possible that this problem doesn't exist on
PulseAudio 5.0 anymore (I didn't explicitly retest it).
The check is general and works for all push based AOs. For pull based
AOs, this can't happen as pull.c implements all the logic correctly.
This should probably be an AO function, but since the playloop still has
some strange stuff (using the buffered_audio variable instead of calling
ao_get_delay() directly), just leave it and make it more explicit.
This collects statistics and other things. The option dumps raw data
into a file. A script to visualize this data is included too.
Litter some of the player code with calls that generate these
statistics.
In general, this will be helpful to debug timing dependent issues, such
as A/V sync problems. Normally, one could argue that this is the task of
a real profiler, but then we'd have a hard time to include extra
information like audio/video PTS differences. We could also just
hardcode all statistics collection and processing in the player code,
but then we'd end up with something like mplayer's status line, which
was cluttered and required a centralized approach (i.e. getting the data
to the status line; so it was all in mplayer.c). Some players can
visualize such statistics on OSD, but that sounds even more complicated.
So the approach added with this commit sounds sensible.
The stats-conv.py script is rather primitive at the moment and its
output is semi-ugly. It uses matplotlib, so it could probably be
extended to do a lot, so it's not a dead-end.
This was accidentally completely destroyed with commit 24f1878e. I
didn't notice it when testing, because forward seeking still worked
mostly.
The issue was that dvd_seek_to_time() actually called stream_seek(),
which was supposed to call the byte-level seek function dvd_seek(). So
we have to restore this function, and replace all generic stream calls
with stream_dvd.c internal ones. This also affects stream->pos (now a
random number as far as stream_dvd.c is concerned) and stream_skip().
The recent change of waking up the playback thread using a wakeup pipe
doesn't work on windows, because windows is horrible. So use a condition
variable instead to wake up the thread. To make things worse, jackaudio
is also horrible and "disallows" the use of mutexes, so all we can do is
implementing a half-solution that is not race condition free.
It would probably better to give up on this lock-free crap in the pull-
API audio path.
Mostly untested.
The audio subsystem now wakes up the playback thread explicitly, and we
don't need this anymore.
It still could cause dropouts and such if there are bugs in the recently
introduced audio changes, so this is a thing to watch out for.
Same change as in e2184fcb, but this time for pull based AOs. This is
slightly controversial, because it will make a fast syscall from e.g.
ao_jack. And according to JackAudio developers, syscalls are evil and
will destroy realtime operation. But I don't think this is an issue at
all.
Still avoid locking a mutex. I'm not sure what jackaudio does in the
worst case - but if they set the jackaudio thread (and only this thread)
to realtime, we might run into deadlock situations due to priority
inversion and such. I'm not quite sure whether this can happen, but I'll
readily follow the cargo cult if it makes hack happy.
I'm not quite sure why ao_pulse needs this. It was broken when a thread
to fill audio buffers was added to AO - the pulseaudio callback was
waking up the playback thread, not the audio thread. But nobody noticed,
so it can't be very important. In any case, this change makes it wake up
the audio thread instead (which in turn wakes up the playback thread if
needed).
And also add a function ao_need_data(), which AO drivers can call if
their audio buffer runs low.
This change intends to make it easier for the playback thread: instead
of making the playback thread calculate a timeout at which the audio
buffer should be refilled, make the push.c audio thread wakeup the core
instead.
ao_need_data() is going to be used by ao_pulse, and we need to
workaround a stupid situation with pulseaudio causing a deadlock because
its callback still holds the internal pulseaudio lock.
For AOs that don't call ao_need_data(), the deadline is calculated by
the buffer fill status and latency, as before.
qscale export has been completely removed from Libav 10, and FFmpeg has
an alternative API, so this code does nothing and only causes
deprecation warnings on Libav.
And slightly adjust the semantics of MPV_EVENT_PAUSE/MPV_EVENT_UNPAUSE.
The real pause state can now be queried with the "core-idle" property,
the user pause state with the "pause" property, whether the player is
paused due to cache with "paused-for-cache", and the keep open event can
be guessed with the "eof-reached" property.
This property is set to "yes" if playback was paused due to --keep-open.
The change notification might not always be perfect; maybe that should
be improved.
Currently this is (probably) equivalent to "paused-for-cache", but the
latter is a bit special, while this new property is a bit more general.
One case where they might actually be different is dvdnav menus, but I
haven't checked.
Also add property change notifications for these two properties.
We were relying on vsscript_freeScript() to take care of proper
termination. But it doesn't do that: it doesn't wait for the filters to
finish and exit at all. Instead, it just destroys all objects, which
causes the worker threads to crash sometimes.
Also, we're supposed to wait for the frame callback to finish before
freeing the associated node.
Handle this by explicitly waiting as far as we can. Probably fixes
crashes on seeking, although VapourSynth itself might also need some
work to make this case completely stable.
It's a bit strange to allow this, so get rid of it.
This probably breaks a bunch of user config files.
The client API still allows setting them with MPV_FORMAT_FLAG with a
value of 1 (i.e. true), but I guess this is tolerable.
The most user visible change is that "420p" is now displayed as
"yuv420p". This is what FFmpeg uses (almost), and is also less confusing
since "420p" is often confused with "420 pixels vertical resolution".
In general, we return the FFmpeg pixel format name. We still use our own
old mechanism to keep a list of exceptions to provide compatibility for
a while.
Also, never return NULL for image format names. If the format is unset
(0/IMGFMT_NONE), return "none". If the format has no name (probably
never happens, FFmpeg seems to guarantee that a name is set), return
"unknown".
This re-allows the previous behaviour of being able to reencode with
metadata removed, which is useful when encoding "inconsistently" tagged
data for a device/player that shows file names when tags are not
present.
Before this commit, the filter attempted to keep the vsscript state
(p->se) even when the script was reloaded. Change it to destroy the
script state too on reloading. Now no workaround for LoadPlugin is
necessary, and this also fixes a weird theoretical race condition when
destroying and recreating the mpv source filter.
I hate tabs.
This replaces all tabs in all source files with spaces. The only
exception is old-makefile. The replacement was made by running the
GNU coreutils "expand" command on every file. Since the replacement was
automatic, it's possible that some formatting was destroyed (but perhaps
only if it was assuming that the end of a tab does not correspond to
aligning the end to multiples of 8 spaces).
This is a read-only property that uses VFCTRL_GET_METADATA
to retrieve mp_tags metadata from a filter specified by label
Signed-off-by: wm4 <wm4@nowhere>
These playlist parsers are all what's left from the old mplayer playlist
parsing code. All of it is old code that does little error checking; the
type of C string parsing code that gives you nightmare.
Some playlist parsers have been rewritten and are located in
demux_playlist.c. The removed formats were not reimplemented. ASX and
SMIL use XML, and since we don't want to depend on a full blown XML
parser, this is not so easy. Possibly these formats could be supported
by writing a very primitive XML-like lexer, which would lead to success
with most real world files, but I haven't attempted that. As for NSC, I
couldn't find any URL that worked with MPlayer, and in general this
formats seems to be more than dead.
Move playlist_parse_file() to playlist.c. It's pretty small now, and
basically just opens a stream and a demuxer. No use keeping
playlist_parser.c just for this.
mpv was resizing to the same size before it went to fullscreen, we don't need to schedule a resize because the compositor will send a configure event with the new dimensions and thats when we should do it.
Basically a cheap hack to fix that the --msgmodule prefix will cause an
unwanted linebreak by making the line too long.
Suggested by Hamuko in github issue #710.
Fixes#710.
This is needed if you want to reimplement the status line in lua
I could only test drop-frame-count because I didn't find an easy way to
trigger paused-for-cache and total-avsync-change
Signed-off-by: wm4 <wm4@nowhere>
Mainly meant to apply simple VapourSynth filters to video at runtime.
This has various restrictions, which are listed in the manpage.
Additionally, this actually copies video frames when converting frame
references from mpv to VapourSynth, and a second time when going from
VapourSynth to mpv. This is inefficient and could probably be easily
improved. But for now, this is simpler, and in fact I'm not sure if
we even can references VapourSynth frames after the core has been
destroyed.
Commit 2c2c1203 sorted the output of --list-options, but the same code
ias also used for listing sub-options, such as --vo=scale:help. For sub-
options, the order actually matters.
Until now, --list-options printed options in random order. There
literally wasn't any logic in its order, they just appeared as they were
declared. So just sort them.
Note that we can't sort them in advance, because for certain things
internal to m_config, the order actually matters.
Also we're using strcasecmp(), which is bad (locale dependent), but this
is output intended for human consumption, so it's not a problem.
This used to display the property type, but it was not always correct or
even available. The way the property mechanism works, we can know this
only at runtime.
Otherwise, the client API user could not know why playback was stopped.
Regarding the fact that 0 is used both for normal EOF and EOF on error:
this is because mplayer traditionally did not distinguish these, and in
general it's hard to tell the real reason. (There are various weird
corner cases which make it hard.)
Although this is something really basic, Lua's standard library doesn't
provide anything like this. Probably because there are too many ways to
do it right or wrong.
This code tries to be really careful when dealing with mixed
arrays/maps, e.g. when a table has integer keys starting from 1, making
it look like an array, but then also has other keys.
The stats were retrieved and written on every encode call, instead of
every encode call that actually returned a packet. ffmpeg.c also does it
this way, so it must be "more correct". Fixes 2-pass encoding.
Our own tables have size for only 8 chars, so these sequences must be
rejected. It seems strings of length 8 are still ok, because the code
uses memcmp and not strcmp, so still allow these.
Based on mplayer-svn commit r37129.
I have some doubts that short reads are even allowed/
possible for /dev/js*, does someone know for sure?
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@37132 b3059339-0415-0410-9bf9-f77b7e298cf2
We needed this because the OSD rendering path used GBRP for RGB
rendering, and not all swscale versions supported this conversion. But
recently we've dropped support for very old ffmpeg/libav versions, so
this isn't needed anymore.
This was broken at some unknown point (even before the recent cache
changes). There are several problems:
- stream_dvd returning a random stream position, confusing the cache
layer (cached data and stream data lost their 1:1 corrospondence by
position)
- this also confused the mechanism added with commit a9671524, which
basically triggered random seeking (although this was not the only
problem)
- demux_lavf requesting seeks in the stream layer, which resulted in
seeks in the cache or the real stream
Fix this by completely removing byte-based seeking from stream_dvd. This
already works fine for stream_dvdnav and stream_bluray. Now all these
streams do time-based seeks, and pretend to be infinite streams of data,
and the rest of the player simply doesn't care about the stream byte
positions.
resize_cache() checks the size itself and clamps the size to the valid
range if necessary, so we don't need these checks. In fact, the checks
are different. Also, output the cache size after clamping, instead of
before.
Use NtQueryVolumeInformationFile instead of GetDriveType for detecting
remote filesystems on Windows. This has the advantage of working
directly on the file handle instead of needing a path and it works
unmodified in Cygwin where the previous code wouldn't understand Cygwin
paths or symlinks.
There is some risk in using NtQueryVolumeInformationFile, since it's an
internal function and its behaviour could change at any time or it could
be removed in a future version of Windows, however it's documented[1] in
the WDK and it's used successfully by Cygwin, so it should be fine. If
it's removed, the code should fail gracefully by treating all files as
local.
[1]: http://msdn.microsoft.com/en-us/library/windows/hardware/ff567070.aspx
Signed-off-by: wm4 <wm4@nowhere>
Merge the cache_read function into cache_fill_buffer, since there's
not much reason to keep them separate. Also, simply call read_buffer()
to see if there's any readable data, instead of checking for the
condition manually.
The only tricky part is keeping the cache contents, which is made simple
by allocating the new cache while still keeping the old cache around,
and then copying the old data.
To explain the "Don't use this when playing DVD or Bluray." comment: the
cache also associates timestamps to blocks of bytes, but throws away the
timestamps on seek. Thus you will experience strange behavior after
resizing the cache until the old cached region is exhausted.
The only difference is that the MP_DBG message is not printed anymore if
the current user read position is outside of the current cache range.
(In order to handle seek_limit==0 gracefully in the normal case of
linear reading, change the comparison from ">=" to ">".)
Until now, this could never happen, because new data was simply always
appended to the end of the cache. But for making stream cache resizing
easier, doing it this way seems advantageous. It also makes it harder to
make the internal state inconsistent. (Before this change it could
happen that cache and stream position went out of sync if the read
position was adjusted "inappropriately".)
Until now, cache_read() (which calls read_buffer()) could return short
reads. This was a simplification allowed by the stream interface. But
for cache resizing, it will be more practical to make read_buffer() do
a full read.
Seems like a good idea. One possible bad effect would be slowing down
uncached controls, but they're already slow. The good thing is that
many controls make intrusive changes to the stream (at least controls
which do write accesses), so the cached parameters should be updated.
Some of these property implementations already send notifications on
their own, but most don't. This takes care of them.
Of course this still doesn't handle all propertry changes - this is
impossible without special-casing each property that can change on its
own.
This might be a good idea in order to prevent queuing a frame too far in
the future (causing apparent freezing of the video display), or dropping
an infinite number of frames (also apparent as freezing).
I think at this point this is most of what we can do if the vdpau time
source is unreliable (like with Mesa). There are still inherent race
conditions which can't be fixed.
The strange thing about this code was the shift parameter of the
prev_vs2 function. The parameter is used to handle timestamps before the
last vsync, since the % operator handles negative values incorrectly.
Most callers set shift to 0, and _usually_ pass a timestamp after the
last vsync. One caller sets it to 16, and can pass a timestamp before
the last timestamp.
The mystery is why prev_vs2 doesn't just compensate for the % operator
semantics in the most simple way: if the result of the operator is
negative, add the divisor to it. Instead, it adds a huge value to it
(how huge is influenced by shift). If shift is 0, the result of the
function will not be aligned to vsyncs.
I have no idea why it was written in this way. Were there concerns about
certain numeric overflows that could happen in the calculations? But I
can't think of any (the difference between ts and vc->recent_vsync_time
is usually not that huge). Or is there something more clever about it,
which is important for the timing code? I can't think of anything
either.
So scrap it and simplify it.
vo_vdpau used a somewhat complicated and fragile mechanism to convert
the vdpau time to internal mpv time. This was fragile as in it couldn't
deal well with Mesa's (apparently) random timestamps, which can change
the base offset in multiple situations. It can happen when moving the
mpv window to a different screen, and somehow it also happens when
pausing the player.
It seems this mechanism to synchronize the vdpau time is not actually
needed. There are only 2 places where sync_vdptime() is used (i.e.
returning the current vdpau time interpolated by system time).
The first call is for determining the PTS used to queue a frame. This
also uses convert_to_vdptime(). It's easily replaced by querying the
time directly, and adding the wait time to it (rel_pts_ns in the patch).
The second call is pretty odd: it updates the vdpau time a second time
in the same function. From what I can see, this can matter only if
update_presentation_queue_status() is very slow. I'm not sure what to
make out of this, because the call merely queries the presentation
queue. Just assume it isn't slow, and that we don't have to update the
time.
Another potential issue with this is that we call VdpPresentationQueueGetTime()
every frame now, instead of every 5 seconds and interpolating the other
calls via system time. More over, this is per video frame (which can be
portantially dropped, and not per actually displayed frame. Assume this
doesn't matter.
This simplifies the code, and should make it more robust on Mesa. But
note that what Mesa does is obviously insane - this is one situation
where you really need a stable time source. There are still plenty of
race condition windows where things can go wrong, although this commit
should drastically reduce the possibility of this.
In my tests, everything worked well. But I have no access to a Mesa
system with vdpau, so it needs testing by others.
See github issues #520, #694, #695.
This turned out ridiculously complex. I think it will have to be
simplified some day. Main reason for the complexity are:
- filtering properties by forcing clients to observe individual
properties explicitly
(to avoid spamming clients with changes they don't want)
- optional retrieval of property value with the notification
(the basic idea was that this is more user friendly)
- allowing to the client to specify a format in which the value
should be retrieved
(because if a property changes its type, the client API couldn't
convert it properly, and compatibility would break)
I don't know yet which of these are important, and everything could
change. In particular, the interface and semantics should be adjusted
to reduce the implementation complexity.
While I consider the API complete, there could (and probably will) be
bugs left. Also while the implementation is complete, it's inefficient.
The complexity of the property matching is O(a*b*c) with a clients,
b observed properties, and c properties changing at once. I threw away
an earlier implementation using bitmasks, because it was too unwieldy.
Remove the use of mp_ring and use a simple array and a bunch of
variables instead. This is way less awkwad.
The change in reserve_reply fixes incorrect tracking of free events.
Now they can be paused and resumed.
Since pausing and disabling the timer is essentially the same underlying
operation, we also just provide one method for it.
mp.cancel_timer probably still works, but I'm considering this
deprecated, and it's removed from the manpage. (We didn't have a release
with this function yet, so no formal deprecation.)
This commit adds support for automatic selection of color profiles based on
the display where mpv is initialized, and automatically changes the color
profile when display is changed or the profile itself is changed from
System Preferences.
@UliZappe was responsible with the testing and implementation of a lot of this
commit, including the original implementation of `cocoa_get_icc_profile_path`
(See #594).
Fixes#594
The --ass-styles option is implemented by calling ass_read_styles().
This function can take a codepage (so libass will use iconv to convert
it). This was implemented before our --subcp option was changed, and
this code was not updated. Now libass fails opening iconv, because
--subcp is not always (and not by default) a valid iconv codepage.
Just always pass NULL, which means the file passed to --ass-styles must
be in UTF-8. The --ass-styles option is a fringe option anyway (and will
destroy your subtitles), so having codepage support for it isn't
important at all.
Change the type of the property from a string list (alternating
key/value entries) to a map. Using the client API, this will return
MPV_FORMAT_NODE_MAP, while Lua mp.get_property_native returns a
dictionary-like table.
The function tag_property() in command.c passed a key action with empty
path to m_property_read_list. This is normally not valid, because key
actions are supposed to access sub-paths. But it's kind of inconvenient
to check for this case in tag_property(). So make it valid by providing
a m_property_unkey() function, which turns a key access to "" into a
top-level action.
We've just checked whether a sub-path started with "name/", but that
changes behavior whether the property name has a trailing '/' or not.
Using a helper function to split of path components avoids this problem.
Also fix a format string mistake in a log call using it.
I wonder if this code shouldn't use FormatMessage, but it looks kind
of involved [1], so: no, thanks.
[1] http://support.microsoft.com/kb/256348/en-us
There are two kind of encryption for Blu-ray disc, AACS and BD+,
and both of them can be checked through BLURAY_DISC_INFO object.
This commit makes the bluray and bdnav streams refuse playback
if AACS/BD+ is detected and decryption is failed.
This played the file at a wrong sample rate if the rate was out of
certain bounds.
A comment says this was for the sake of libaf/af_resample.c. This
resampler has been long removed. Our current resampler
(libav/swresample) checks supported sample rates on reconfiguration, and
will error out if a sample rate is not supported. And I think that is
the correct behavior.
The angles should be set and queried only if a valid title is
selected. Also, in navigation mode, there are some limitations
which make it impossible to query current title/angle.
This commit introduces new stream protocols: bdnav(and others).
bdnav stream shares lots of codes with original bluray stream, so
it's not separated in different source file.
Major difference from bluray is that bdnav does not support longest
title because there is no way to query that information.
bdnav://menu and bdnav://first correspond to top menu title and
first play title respectively, though they often point same title.
Also, binary position based seeking has been removed, because it
didn't have no point.
Reduce most dependencies on struct mp_csp_details, which was a bad first
attempt at dealing with colorspace stuff. Instead, consistently use
mp_image_params.
Code which retrieves colorspace matrices from csputils.c still uses this
type, though.
This is pretty obscure, so it didn't matter much. It still breaks
switching output levels at runtime, because the video output is not
reinitialized with the new params.
This was accidentally broken in commit b72ba3f7. I somehow made the
wild assumption that replaygain adjusted the volume relative to 0%
instead of 100%.
The detach suboption was similarly broken.
This was actually supposed to be removed with pull reuqest #671, but
I accidentally re-added it with a rebasing mistake.
This probably also coincidentally fixes compilation with older
libbluray (issue #672).
Use bd_get_playlist_info() instead of bd_get_title_info(). The
previous implementation couldn't query current playlist and this
made it impossible to call bd_get_playlist_info() which is more
desirable than bd_get_title_info() because, for Blu-rays, playlist
is the unit of playback not title. This commit fixes that.
The cost of calling bd_get_title_info() is quite expensive and
requires lots of CPU usage. Using BD_EVENT_PLAYLIST and
BD_EVENT_TITLE, it's possible to cache BLURAY_TITLE_INFO object for
current title and BD_EVENT_ANGLE handler caches current angle. In
my test case, with this commit, CPU usage can be saved about 15-20%.
demux_mf.c explicitly checks for the stream type to check whether images
are opened via pattern (mf://..., i.e. stream_mf.c) or directly. Of
course the stream type is not set to STREAMTYPE_MF if the stream is
wrapped through the cache, so it tried to open the pattern directly as
file, which failed.
Fix this by disabling caching for mf://. The cache doesn't make sense
here anyway, because each file is opened and closed every frame (perhaps
to avoid memory bloat).
It's possible that MPContext has a chapter list, but the demuxer
doesn't. In this case, accesing the chapter-metadata property would
lead to invalid accesses.
(This fixes the out of bound access, but in theory, the returned data
can still be incorrect, since MPContext chapters don't need to map
directly to demuxer chapters.)
This cd_info_t struct was practically unused. The only thing it did was
storing the track name of the form "Track %d" in a very roundabout way.
Remove it. (It made more sense when there was still CDDB support.)
Don't use an integer division to get the time, since that would round on
second boundaries. Also round up the time by sector size. Seeking rounds
down due to alignment constraints, but if we round up the time, we can
make it land on the exact destination sector.
This fixes that the track change code printed the previous track when
seeking by chapter.
This was usually handled at the end of the switch statement, so if
something returns from the function before that, the event has to be
freed explicitly.
Previous implementation updated video resolution when highlight
event was given. However, this may not work if highlight event
is given before video size is queried.
This commit adds checking routine into rendering function, too.
dvdnav.c did not handle event in regular sequence. Usually this
does not make any trouble except around MP_NAV_EVENT_RESET_ALL.
Those events should be handled in regular sequence. If they're
mixed, it can make wrong result.
For instance, MP_NAV_EVENT_HIGHLIGHT right after
MP_NAV_EVENT_RESET_ALL should not be ignored but it might be
because MP_NAV_EVENT_RESET_ALL makes the demuxer reloaded and osd
hidden.
And consistently use MP_NOPTS_VALUE as error value for the users of this
function. This is better than using -1, especially because negative
values can be valid timestamps.
Instead of comparing the current chapter every time, set the playback
end timestamp to the chapter end. Likewise, don't execute an extra seek
for the start chapter.
Maybe we could also use the timeline facility to restrict playback to
the given chapter range, but this would be strange when using
--chapter=N to start playback at a given chapter. Then you couldn't seek
back, which is possibly not what the user wants.
Instead, always use the mpctx->chapters array. Before this commit, this
array was used only for ordered chapters and such, but now it's always
populated if there are chapters.
Stream-level chapters (like DVD etc.) did potentially not have
timestamps for each chapter, so STREAM_CTRL_SEEK_TO_CHAPTER and
STREAM_CTRL_GET_CURRENT_CHAPTER were needed to navigate chapters. We've
switched everything to use timestamps and that seems to work, so we can
simplify the code and remove this old mechanism.
Report the time for each chapter (tracks are treated as chapters). This
allows us to get rid of the "old" chapter mechanism, and also behaves
better with the frontend.
This makes assumptions about the audio formats, but that format is
hardcoded anyway in the rawaudio demuxer defaults (and always was).
Libav 10 was released, so we can enable testing the stable Libav version
again.
FFmpeg 2.2 was also released, but since we still support 2.1.4, we stick
with the older version. This is better for testing.
There were some bad interactions with the OSC.
For one, dragging the OSC bar, and then moving the mouse outside of the
OSC (while mouse button still held) would suddenly initiate window
dragging. This was because win_drag_button1_down was not reset when
sending a normal mouse event, which means the window dragging code can
become active even after we've basically decided that the preceding
click didn't initiate window dragging.
Second, dragging the window and clicking on the OSC bar after that did
nothing. This was because no mouse button up event was sent to the core,
even though a mouse down event was sent. So make sure the key state is
erased with MP_INPUT_RELEASE_ALL.
We don't check whether the WM supports _NET_WM_MOVERESIZE_MOVE, but
if it doesn't, nothing bad happens. There might be a race condition
when pressing a button, and then moving the mouse and releasing the
button at the same time; then the WM might get the message to initiate
moving the window after the mouse button has been released, in which
case the result will probably be annoying. This could possibly be fixed
by sending _NET_WM_MOVERESIZE_CANCEL on button release, but on the
other hand, we probably won't receive a button release event in this
situation, so ignore this problem.
The dragging is initiated only when moving the mouse pointer after a
click in order to reduce annoying behavior when the user is e.g.
doubleclicking.
Closes#608.
The title for stream_bluray DID start from 1 and I misunderstood
that it started from 0 because mpv accepted bd://0 as a proper
argument. In fact, 0 title was an alias for the longest title but
it was not handled as a special value. This commit fixes these
behavious. 'disc-title' property for Blu-ray now starts from 0 and
the default title can be specified by 'longest' title just like
stream_dvdnav: bd://longest. Of course, 'longest' can be omitted.
This commit makes 'disc-title' property writable using
STREAM_CTRL_SET_CURRENT_TITLE. This commit also contains
implementation of STREAM_CTRL_SET_CURRENT_TITLE for stream_bluray.
Currently, 'disc-title' is writable only for stream_dvdnav and
stream_bluray and stream_dvd is not supported.
This commit makes 'disc-title' properties for DVDs start from 0.
There was an inconsistency around 'disc-title' property between
DVDs (from 1) and Blu-rays (from 0). This fixes#648.
Signed-off-by: wm4 <wm4@nowhere>
This check incorrectly passed on Cygwin. While Cygwin has the
statfs.f_type field, it contains the volume's FileSystemAttributes,
which aren't useful for detecting network mounts.
"enable-osc" will make the OSC appear at any time (although it'll
quickly disappear again if the mouse is not inside the OSC). "disable-
osc" will make it disappear permanently.
Also, if the OSC is visible, force remap the DEL key to make the OSC
disappear.
Change script_message to broadcast the message to all clients. Add a new
script_message_to command, which does what the old script_message
command did.
This is intended as simplification, although it might lead to chaos too.
VAAPI has some ambiguous image formats, like VA_FOURCC_I420,
VA_FOURCC_IYUV, VA_FOURCC_YV12 (the latter exactly the same as the first
two, just with swapped planes). There is potentially a problem when one
specific VAAPI format was picked, and converting it to a mpv format and
back to a VAAPI FourCC would result in a numerically different format
(even if it's actually the same). Then it could e.g. happen that
functions like va_surface_upload() reallocate the underlying VAImage,
which would be inefficient. Change the code so that this can't happen.
(Probably not a problem in practice with the current VAAPI usage.)
It's not really needed to be public. Other code can just use mp_image.
The only disadvantage is that the other code needs to call an accessor
to get the VASurfaceID.
Although I at first thought it would be better to have a separate
implementation for hwaccels because the difference to software images
are too large, it turns out you can actually save some code with it.
Note that the old implementation had a small memory management bug. This
got painted over in commit 269c1e1, but is hereby solved properly.
Also note that I couldn't test vf_vavpp.c (due to lack of hardware), and
I hope I didn't accidentally break it.
The plan is to get rid of the custom VAAPI and possibly VDPAU surface
allocators.
Add custom surface allocation, because hwaccel surfaces are allocated
completely differently from software surfaces.
Add optional LRU allocation, which is (probably) helpful for hwaccel,
but (probably) less optimal for software surfaces.
mp_image_pool_get_no_alloc() is specifically for VAAPI, which can't
allocate new decoder surfaces after decoder init.
They were used by ancient libavcodec versions. This also removes the
need to distinguish vdpau image formats at all (since there is only
one), and some code can be simplified.
Image formats used to be FourCCs, so unsigned int was better. But now
it's annoying and the only difference is that unsigned int is more to
type than int.
Instead of doing it on every seek (libavcodec calls get_format on every
seek), reinitialize the decoder only if the video resolution changes.
Note that this may be relatively naive, since we e.g. (or: in
particular) don't check for profile changes. But it's not worse than the
state before the get_format change, and at least it paints over the
current vaapi breakage (issue #646).
Set refcounted_frames, because in some versions of libavcodec mixing the
new AVFrame API and non-refcounted decoding could cause memory
corruption. Likewise, it's probably still required to unref a frame
before calling the decoder.
This follows the (only slowly progressing) plan to replace all internal
video filters with libavfilter.
All what's left in vf_gradfun.c is the weird wrapper around vf_lavfi.c.
I have no tolerance for this garbage anymore. There are tons of issues
with it (see e.g. previous commit), and there is no reason to use it
either. Use Libav git, or Libav 10 when it's released.
This also drops support for earlier FFmpeg release, which have exactly
the same issues as Libav 9. FFmpeg 2.1.4 is still supported, because
it's the latest release, and is reasonably recent. (Although this will
probably also be dropped as soon as FFmpeg 2.2 is released.)
Assumed version table (lots of nonsensical numbers):
FFmpeg 2.1.4 FFmpeg (n2.2-rc2) Libav (v10_beta2)
lavu 52.48.101 52.66.100 53.3.0
lavc 55.39.101 55.52.102 55.34.1
lavf 55.19.104 55.33.100 55.12.0
lsws 2.5.101 2.5.101 2.1.2
lavi 3.90.100 4.2.100 4.2.0
lswr 0.17.104 0.18.100 -
lavr 1.1.0 1.2.0 1.1.0
libpostproc and libavdevice are not interesting.
Following this commit, code needed just to support old Libav versions
will start to be removed.
This "sometimes" crashed when seeking. The fault apparently lies in
libavcodec: the decoder returns an unreferenced frame! This is
completely insane, but somehow I'm apparently still expected to
work this around. As a reaction, I will drop Libav 9 support in the
next commit. (While this commit will go into release/0.3.)
Instead of parsing the ASS file in demux_libass.c and trying to pass the
ASS_Track to the subtitle renderer, just read all file data in
demux_libass.c, and let the subtitle renderer pass the file contents to
ass_process_codec_private(). (This happens to parse full files too.)
Makes the code simpler, though it also relies harder on the (messy)
probe logic in demux_libass.c.
They're strictly DVD-only, so it's better to mark them as such. This
also documentes the "title" (now renamed to "dvd-title") property.
This also avoids collision with the --title option. (Technically, there
was no problem. But it might be confusing for users, since we have a
policy of naming properties and options the same if they refer to the
same underlying functionality.)
This commit adds new property 'title' which indicates current
playing title of disc. This property is useful when using a stream
whose title can be changed during playback, e.g., dvdnav.
This commit provides impelmentation of STREAM_CTRL_GET_NUM_TITLES
for dvdnav stream. Other streams for DVD or Blu-ray are already
provide STREAM_CTRL_GET_NUM_TITLES.
Maybe this should be default. On the other hand, this filter does
something even if the volume is neutral: it clips samples against the
allowed range, should the decoder or a previous filter output garbage.
Currently, both replaygain adjustment and user volume control (if
softvol is enabled) share the same variable. Sharing the variable would
cause especially if --volume is used; then the replaygain volume would
always be overwritten.
Now both gain values are simple added right before doing filtering.
Xlib.h (included from x11_common.h) defines a macro 'Status' as 'int'.
This messed up a bunch of definitions in windows.h and broke the build
in Cygwin. Including windows.h first seems to solve the problem.
This commit also removes the definition of an unused flag.
This adds the options replaygain-track and replaygain-album. If either is set,
the replaygain track or album gain will be automatically read from the track
metadata and the volume adjusted accordingly.
This only supports reading REPLAYGAIN_(TRACK|ALBUM)_GAIN tags. Other formats
like LAME's info header would probably require support from libav.
The main incompatibility was that Libav didn't have av_opt_set_int_list.
But since that function is excessively ugly and idiotic (look how it
handles types), I'm not missing it much. Use an aformat filter instead
to handle the functionality that was indirectly provided by it. This is
similar to how vf_lavfi works.
The other incompatibility was channel handling. Libav consistently uses
channel layouts only, why ffmpeg still requires messing with channel
counts to some degree. Get rid of most channel count uses (and hope
channel layouts are "exact" enough). Only in one case FFmpeg fails with
a runtime check if we feed it AVFrames with channel count unset.
Another issue were AVFrame accessor functions. FFmpeg introduced these
for ABI compatibility with Libav. I refuse to use them, and it's not my
problem if FFmpeg doesn't manage to provide a stable ABI for fields
provided both by FFmpeg and Libav.
And make some additions.
I'm not really sure why we list dependencies that are not ours, but that
of external libraries. Still might be useful, though.
I've tried not to be too detailed (because it's not a reference, just
some guidelines), but it still got relatively long. Also contains
conventions for sending patches.
The window doesn't recieve a WM_LBUTTONUP message after it's dragged,
probably because it's swallowed by the modal loop. To stop the button
from sticking, release it manually when the drag is complete.
Mouse buttons can get stuck down if the button is pressed inside the
video window and released outside. Avoid this by capturing mouse input
when a button is pressed.
Linux also has fstatfs(), and the test relied on certain system include
files being available on BSD, but not on Linux. It would break if Linux
added the missing includes for some reason.
Make it a bit stricter, and check for the struct statfs field the code
needs.
This reads MPV_CLIENT_API_VERSION from the source header, and turns it
into a 3 part version number.
E.g. if MPV_CLIENT_API_VERSION were 0x12abcdef, this would result in
"18.171.773615" (8 bits, 8 bits, 16 bits).
We'll see if this is actually useful, or if it's too clever.
Rename it to --enable-libmpv-shared. The option name didn't really
tell much. When we add the possibility to create a static library,
it would also be bad if that were named --enable-static (because it
would sound like it does what --static-build does).
The volume controls in mpv now affect the session's volume (the
application's volume in the mixer). Since we do not request a
non-persistent session, the volume and mute status persist across mpv
invocations and system reboots.
In exclusive mode, WASAPI doesn't have access to a mixer so the endpoint
(sound card)'s master volume is modified instead. Since by definition
mpv is the only thing outputting audio in exclusive mode, this causes no
conflict, and ao_wasapi restores the last user-set volume when it's
uninitialized.
Due to the COM Single-Threaded Apartment model, the thread owning the
objects will still do all the actual method calls (in the form of
message dispatches), but at least this will be COM's problem rather than
having to set up several handles and adding extra code to the event
thread.
Since the event thread still needs to own the WASAPI handles to avoid
waiting on another thread to dispatch the messages, the init and uninit
code still has to run in the thread.
This also removes a broken drain implementation and removes unused
headers from each of the files split from the original ao_wasapi.c.
ao_wasapi.c was almost entirely init code mixed with option code and
occasionally actual audio handling code. Split most things to
ao_wasapi_utils.c and keep the audio handling code in ao_wasapi.c.
Gets rid of the internal ring buffer and get_buffer. Corrects an
implementation error in thread_reset.
There is still a possible race condition on reset, and a few refactors
left to do. If feasible, the thread that handles everything
WASAPI-related will be made to only handle feed events.
This reverts commit 75dd3ec210.
This broke seeking with ordered chapters in some situations. While
the reverted commit was perfectly fine for playback of normal files,
it overlooked that in the ordered chapters case switching segments
actually reinitialized the audio chain completely, including the
decoder. And decoders still read packets on initialization. We can
restore the original commit as soon as decoders stop doing this.
Apparently the "right" place to initialize the hardware decoder is in
the libavcodec get_format callback.
This doesn't change vda.c and vdpau_old.c, because I don't have OSX, and
vdpau_old.c is probably going to be removed soon (if Libav ever manages
to release Libav 10). So for now the init_decoder callback added with
this commit is optional.
This also means vdpau.c and vaapi.c don't have to manage and check the
image parameters anymore.
This change is probably needed for when libavcodec VDA supports gets a
new iteration of its API.
This updates the logic for the new, somewhat unified behavior of SRGB
and 3DLUT since 34bf9be (not that it was particularly correct even that
change) and checks for the presence of corresponding extensions only in
the cases in which they're needed.
This commit:
- Changes some of the #define and variable names for clarification and
adds comments where appropriate.
- Unifies :srgb and :icc-profile, making them fit into the same step of
the decoding process and removing the weird interactions between both
of them.
- Makes :icc-profile take precedence over :srgb (to significantly reduce
the number of confusing and useless special cases)
- Moves BT709 decompanding (approximate or actual) to the shader in all
cases, making it happen before upscaling (instead of the old 0.45
gamma function). This is the simpler and more proper way to do it.
- Enables the approx gamma function to work with :srgb as well due to
this (since they now share the gamma expansion code).
- Renames :icc-approx-gamma to :approx-gamma since it is no longer tied
to the ICC options or LittleCMS.
- Uses gamma 2.4 as input space for the actual 3DLUT, this is now a
pretty arbitrary factor but I picked 2.4 mainly because a higher pure
power value here seems to produce visually better results with wide
gamut profiles, rather then the previous 1.95 or BT.709.
- Adds the input gamma space to the 3dlut cache header in case we change
it more in the future, or even make it user customizable (though I
don't see why the latter would really be necessary).
- Fixes the OSD's gamma when using :srgb, which was previously still
using the old (0.45) approximation in all cases.
- Updates documentation on :srgb, it was still mentioning the old
behavior from circa a year ago.
This commit should serve to both open up and make the CMS/shader code much
more accessible and less confusing/error-prone and simultaneously also
improve the performance of 3DLUTs with wide gamut color spaces.
I would liked to have made it more modular but almost all of these
changes are interdependent, save for the documentation updates.
Note: Right now, the "3DLUT takes precedence over SRGB" logic is just
coded into gl_lcms.c's compile_shaders function. Ideally, this should be
done earlier, when parsing the options (by overriding the actual
opts.srgb flag) and output a warning to the user.
Note: I'm not sure how well this works together with real-world
subtitles that may need to be color corrected as well. I'm not sure
whether :approx-gamma needs to apply to subtitles as well. I'll need to
test this on proper files later.
Note: As of now, linear light scaling is still intrinsically tied to
either :srgb or :icc-profile. It would be thinkable to have this as an
extra option, :linear-scaling or similar, that could be used with or
without the two color management options.
Assume obtained.samples contains the number of samples the SDL audio
callback will request at once. Then make sure ao.c will set the buffer
size at least to 3 times that value (or more).
Might help with bad SDL audio backends like ESD, which supposedly uses a
500ms buffer.
This obviously doesn't work. It wasn't much of a problem in the past
because most passthrough formats use 2 channels, which is also the
default for downmix.
In general, we don't need to have a large hw audio buffer size anymore,
because we can quickly fill it from the soft buffer.
Note that this probably doesn't change much anyway. On my system (dmix
enabled), the buffer size is only 170ms, and ALSA won't give more. Even
when using a hardware device the buffer size seems to be limited to
341ms.
This AO pretended to support volume operations when in spdif passthrough
mode, but actually did nothing. This is wrong: at least the GET
operations must write their argument. Signal that volume is unsupported
instead.
This was probably a hack to prevent insertion of volume filters or so,
but it didn't work anyway, while recovering after failed volume filter
insertion does work, so this is not needed at all.
Since the addition of the AO feed thread, 200ms of latency (MIN_BUFFER)
was added to all push-based AOs. This is not so nice, because even AOs
with relatively small buffering (e.g. ao_alsa on my system with ~170ms
of buffer size), the additional latency becomes noticable when e.g.
toggling mute with softvol.
Fix this by trying to keep not only 200ms minimum buffer, but also 200ms
maximum buffer. In other words, never buffer beyond 200ms in total. Do
this by estimating the AO's buffer fill status using get_space and the
initially known AO buffer size (the get_space return value on
initialization, before any audio was played). We limit the maximum
amount of data written to the soft buffer so that soft buffer size and
audio buffer size equal to 200ms (MIN_BUFFER).
To avoid weird problems with weird AOs, we buffer beyond MIN_BUFFER if
the AO's get_space requests more data than that, and as long as the soft
buffer is large enough.
Note that this is just a hack to improve the latency. When the audio
chain gains the ability to refilter data, this won't be needed anymore,
and instead we can introduce some sort of buffer replacement function in
order to update data in the soft buffer.
It is possible to have ao->reset() called between ao->pause() and
ao->resume() when seeking during the pause. If the underlying PCM
supports pausing, resuming an already reset PCM will produce an error.
Avoid that by explicitly checking PCM state before calling
snd_pcm_pause().
Signed-off-by: wm4 <wm4@nowhere>
The uint64_t math would cause overflow at long enough system uptimes
(...such as 3 days), and any precision error given by the double math will
be under one milisecond.
One strange issue is that we apparently can't stop the audio API on
audio reset (ao_driver.reset). We could use SDL_PauseAudio, but that
doesn't specify whether remaining audio is dropped. We also could use
SDL_LockAudio, but holding that over a long time will probably be bad,
and it probably doesn't drop audio. This means we simply play silence
after a reset, instead of stopping the callback completely. (The
existing code ran into an underrun in this situation.)
The delay estimation works about the same. We simply assume that the
callback is locked to audio timing (like ao_jack), and that 1 callback
corresponds to 1 period. It seems this (removed) code fragment assumes
there 1 one period size delay:
// delay subcomponent: remaining audio from the next played buffer, as
// provided by the callback
buffer_interval += callback_interval;
so we explicitly do that too.
Apparently, this is always _really_ monotonic, despite what the Linux
manpages say. So this should be much better than gettimeofday(). (At
times there were kernel bugs which broke the monotonic property.)
From the perspective of the player, time can still be discontinuous
(you could just stop the process with ^Z), but at least it's guaranteed
to be monotonic without further hacks required.
Also note that clock_gettime() returns the time in nanoseconds. We want
microseconds only, because that's the unit we chose internally. Another
problem is that nanoseconds can wrap pretty quickly (less than 300 years
in 63 bits), so it's just better to use microseconds. The devision won't
make the code that much slower (compilers can avoid a real division).
Note: this expects that the system provides clock_gettime() as well as
CLOCK_MONOTONIC. Both are optional according to POSIX. The only system
I know which doesn't have these, OSX, has seperate timer code anyway,
but I still don't know whether more obscure (yet supported) platforms
have a problem with this, so I'm playing safely. But this still expects
that CLOCK_MONOTONIC always works at runtime if it's defined.
This is probably "safer". Without it, we will play 1 sample, because the
logic was written in a way to decode 1 sample if audio is paused. 1
sample usually will initialize the audio PTS, but not play any real
audio. Also see previous commit.
In ancient times, this actually used 1 byte (instead of 1 sample), so
clearly no sample was written, unless the audio was 8-bit mono.
Remove the ao_buffer_playable_samples field. This contained the number
of samples that fill_audio_out_buffers() wanted to write to the AO (i.e.
this data was supposed to be played at some point), but ao_play()
rejected it due to partial fill.
This could happen with many AOs, notably those which align all written
data to an internal period size (often called "outburst" in the AO
code), and the accepted number of samples is rounded down to period
boundaries. The left-over samples at the end were still kept in
mpctx->ao_buffer, and had to be played later.
The reason ao_buffer_playable_samples had to exist was to make sure that
at EOF, the correct number of left-over samples was played (and not
possibly other data in the buffer that had to be sliced off due to
endpts in fill_audio_out_buffers()). (You'd think you could just slice
the entire buffer, but I suspect this wasn't done because the end time
could actually change due to A/V sync changes. Maybe that was the reason
it's so complicated.)
Some commits ago, ao.c gained internal buffering, and ao_play() will
never return partial writes - as long as you don't try to write more
samples than ao_get_space() reports. This is always the case. The only
exception is filling the audio buffers while paused. In this case, we
decode and play only 1 sample in order to initialize decoding (e.g. on
seeking). Actually playing this 1 sample is in fact a bug, but even of
the AO doesn't have period size alignment, you won't notice it. In
summary, this means we can safely remove the code.
Until now, this was always conflated with uninit. This was ugly, and
also many AOs emulated this manually (or just ignored it). Make draining
an explicit operation, so AOs which support it can provide it, and for
all others generic code will emulate it.
For ao_wasapi, we keep it simple and basically disable the internal
draining implementation (maybe it should be restored later).
Tested on Linux only.
Same deal as with the previous commit. We don't lose any functionality,
except for waiting "properly" on audio end, instead of waiting using the
delay estimate.
This removes the ringbuffer management from the code, and uses the
generic code added with the previous commit. The result should be
pretty much the same.
The "estimate" sub-option goes away. This estimation is now always
active. The new code for delay estimation is slightly different, and
follows the claim of the jack framework that callbacks are timed
exactly.
This has 2 goals:
- Ensure that AOs have always enough data, even if the device buffers
are very small.
- Reduce complexity in some AOs, which do their own buffering.
One disadvantage is that performance is slightly reduced due to more
copying.
Implementation-wise, we don't change ao.c much, and instead "redirect"
the driver's callback to an API wrapper in push.c.
Additionally, we add code for dealing with AOs that have a pull API.
These AOs usually do their own buffering (jack, coreaudio, portaudio),
and adding a thread is basically a waste. The code in pull.c manages
a ringbuffer, and allows callback-based AOs to read data directly.
Since the AO will run in a thread, and there's lots of shared state with
encoding, we have to add locking.
One case this doesn't handle correctly are the encode_lavc_available()
calls in ao_lavc.c and vo_lavc.c. They don't do much (and usually only
to protect against doing --ao=lavc with normal playback), and changing
it would be a bit messy. So just leave them.
We want to move the AO to its own thread. There's no technical reason
for making the ao struct opaque to do this. But it helps us sleep at
night, because we can control access to shared state better.
This field will be moved out of the ao struct. The encoding code was
basically using an invalid way of accessing this field.
Since the AO will be moved into its own thread too and will do its own
buffering, the AO and the playback core might not even agree which
sample a PTS timestamp belongs to. Add some extrapolation code to handle
this case.
Use QueryPerformanceCounter to improve the accuracy of
IAudioClock::GetPosition.
While this is mainly for "realtime correctness" (usually the delay is a
single sample or less), there are cases where IAudioClock::GetPosition
takes a long time to return from its call (though the documentation doesn't
define what a "long time" is), so correcting its value might be important in
case the documented possible delay happens.
The lack of device latency made get_delay report latencies shorter than
they should; on systems with fast enough drivers, the delay is not
perceptible, but high enough invisible delays would cause desyncs.
I'm not yet completely sure whether this is 100% accurate, there are
some issues involved when repeatedly pausing+unpausing (the delay might
jump around by several dozen miliseconds), but seeking seems to be
working correctly now.
The player didn't quit when the end of a file was reached. The reason
for this is that jack reported a constant audio delay even when all
audio was done playing. Whether that was recognized as EOF by the player
depended whether the exact value was higher or lower than the player's
threshhold for what it considers no more audio.
get_delay() should return amount of time it takes until the last sample
written to the audio buffer reaches the speaker. Therefore, we have to
track the estimated time when the last sample is done, and subtract it
from the calculated latency. Basically, the latency is the only amount
of time left in the delay, and it should go towards 0 as audio reaches
ths speakers.
I'm not sure if this is correct, but at least it solves the problem. One
suspicious thing is that we use system time to estimate the end of the
audio time. Maybe using jack_frame_time() would be more correct. But
apart from this, there doesn't seem to be a better way to handle this.
For example, consider the case when audio initialization fails. Then the
audio track is deselected. Before this commit, this would have been
equivalent to the user disabling audio. This is bad when multiple files
are played at once (the next file would have audio disabled, even if it
works), or if playback resume is used (if e.g. audio output failed to
initialize, then audio would be disabled when resuming, even if the
system's audio driver was fixed).
Setting string options to strings over the m_option fallback (i.e.
M_PROPERTY_SET_STRING is called if the option type is CONF_TYPE_STRING)
failed. This was because m_option_parse() returns 0. 0 still means
success, but the property code tried to be clever, and considered 0 not
a success in order to disallow setting flags to an emtpy string (which
in turn is allowed, because the command line allows flag options without
parameters).
Fix this by removing the overly clever code.
This could happen when e.g. using the "set" command on options/title (a
string option), and also was a problem for the client API.
Closes#610.
The OSC used significant CPU time while the player was paused. It turned
out that the "tick" event sent during pause is the problem. The OSC
accesses the player core when receiving a tick event, which in turn will
cause the core to send another tick event, leading to infinite feedback.
Fix this by sending an idle tick only every 500ms. This is not very
proper, but the idea behind the tick event isn't very clean to begin
with (and the OSC should use timers instead).
The playloop usually waits in select(), using a timeout needed for
refilling audio and video buffers. This means the client API needs
a separate mechanism to interrupt the select() call. This mechanism
exists, but I forgot to use it. This commit fixes it.
If it works, this will make the client API react faster, epsecially
in audio-only mode. If video is enabled, the reaction time is capped
to 50ms (or somewhat faster if the framerate is >20 fps), because
the playloop stops reacting to anything in order to render and time
the next video frame. (This will be fixed later by moving the VO
to its own thread.)
--ass-style-override=force now attempts to override the 'Default' style.
May or may not work. In some situations it will work, but also mess up
seemingly unrelated things like signs typeset with ASS.
luaL_loadstring(), which was used until now, uses the start of the Lua
code itself as chunk name. Since the chunk name shows up even with
runtime errors triggered by e.g. Lua code loaded from user scripts, this
looks a but ugly. Switch to luaL_loadbuffer(), which is almost the same
as luaL_loadstring(), but allows setting a chunk name.
Will be helpful to track down strange wait times and such issues, as
well when you have develop something timing related. (Then you may print
timestamps in your debug output, and the --msgtime timestamps will help
giving context.)
Sending an asynchronous request and then calling mpv_destroy() would
crash the player when trying to send the reply to the removed client.
Fix this by waiting until all remaining replies have been sent.
The step argument for "add volume <step>" was ignored until now. Fix it.
There is one problem: by defualt, "add volume" should use the value set
with --volstep. This value is 3 by default. Since the default volue for
the step argument is always 1 (and we don't really want to make the
generic code more complicated by introducing custom step sizes), we
simply multiply the step argument with --volstep to keep it compatible.
The --volstep option should probably be just removed in the future.
The value range is 0-100, so fractional values don't make much sense.
But the underlying data type is probably float to avoid getting "stuck"
when doing small volume increments. So step this around and pretend it's
an integer just on display.
The previous version of the gamma suboption was pretty useless. It could
be used to disable delayed gamma enabling, which is a mechanism to avoid
having to adjust gamma in the shader by default.
Repurpose the suboption and allow setting an exact gamma value with it.
You can already override gamma with the --gamma option as well as the
gamma input property, but these use a weird curve to create the
impression of a linear perceived brightness change when changing the
value. This suboption now allows setting an exact gamma value.
This used to be absolute colorimetric, but relative colorimetric is a
saner default due to the arguments presented in issue #595.
A short summary: In general it doesn't affect much because our eyes
adapt to the white point either way, but if running in windowed mode it
would make the whites seem inconsistent/tinted. For fullscreen
projection it's also undesirable since it reduces the dynamic range
without much benefit (again, since our eyes adapt either way) and it
also breaks calibration against ambient lighting.
This shouldn't change much, since most profile types that aren't 3DLUTs
aren't capable of either of those transforms, and most displays are
calibrated against D65 (same as BT.709 source) either way.
Lua doesn't distinguish between arrays and maps on the language level;
there are just tables. Use metatables to mark these tables with their
actual types. In particular, it allows distinguishing empty arrays from
empty tables.
When passing a very large timeout to mpthread_cond_timed_wait(), the
calculations could overflow, setting tv_sec to a negative value, and
making the pthread_cond_timed_wait() call return immediately. This
accidentally made Lua support poll and burn CPU for no reason.
The existing overflow check was ineffective on 32 bit systems. tv_sec is
usually a long, so adding INT_MAX to it will usually not overflow on 64
bit systems, but on 32 bit systems it's guaranteed to overflow. Simply
fix by clamping against a relatively high value. This will work until 1
week before the UNIX time wraps around in 32 bits.
Always map MPV_FORMAT_STRING to setting property value directly through
M_PROPERTY_SET_STRING, instead of trying to go through
M_PROPERTY_SET_NODE.
This treats a direct MPV_FORMAT_STRING query differently from a
MPV_FORMAT_STRING wrapped in a mpv_node. This was already the case in
mpv_get_property(). The reason for all this is that mpv_node is supposed
to be the exact type, while a direct MPV_FORMAT_STRING goes through all
possible conversions.
Not sure if these semantics are good.
This is a bit weird: m_option_string types (i.e. char*) can be NULL. But
they're supposed to be treated just like empty strings. So don't make
the m_option_type.print function return NULL for these values. Returning
NULL would mean failure.
This didn't matter much before, but was quite visible through the client
API.
1. Cannot set option after initialized: it seems that this bug has
existed since libmpv was introduced first. Maybe just a typo.
2. Crash when setting property with native format: mpv_set_property
just causes a crash when using a native format. I found an invalid
casting and fixed it.
3. Wrong error value for mpv_get_property: when an error occurred,
mpv_get_property always returns wrong format error because every
error for property except M_PROPERTY_NOT_IMPLEMENTED is just ignored.
Signed-off-by: wm4 <wm4@nowhere>
Closes pull request #593. Does not incldue the first fix, which was not
correct. The underlying bug will be fixed by a later commit.
Commit message extracted from pull request and slightly edited.
E.g. binding MOUSE_BTN0 always used the user defined binding. While it
is ok that the user can override mouse_move and mouse_leave (for
whatever reasons), we want to strictly override the bindings when input
is sent to the OSC itself.
Regression since 03624a1.
This lowers the number of data stored in the resume config a bit
further, because some properties can't be read at program start and when
e.g. the VO wasn't created yet.
Some fields still need to be read from a property (actually only
"volume-restore-data", a hack to save the full volume information). So
abuse the "options/" property, and make use of the fact that changing
things at runtime also changes the options.
This is approximate: we read each option value on program start
(before starting playback of a file), and when writing the resume
config, compare each value to the current state. This also means
when a value is changed and then changed back, it's not stored. In
particular, option values set in config files and on the command
line are considered the default.
This should help reducing the numbers of options overridden by the
resume config. If too much is overridden, it becomes an inconvenience,
because changes in config files will apparently have no effect when
resuming a file.
Also see github issue #574.
This might be helpful if we ever want cascading config files. Also, we
will probably need it if we change the default input.conf bindings, and
want to provide compatibility input.conf files.
This created an essentially empty config file. This is not really
needed and probably causes more trouble than it solves (such as
littering the home directory with crap), so get rid of it.
This was broken by commit bb6b543812. Note that the original pull
request was fine, but it was broken by my own stupidity when I was
"improving" it.
The problem is that the new loadfile argument was not considered
optional anymore after my changes. The original pull request did handle
this by setting .defval to a dummy value, but I removed that part.
Fix it again by introducing a flag that designates that the parameter is
optional. (I didn't want to add it to m_option.h, because technically,
all options are optional, and it's not possible to have non-optional
options.)
Not sure about this... might redo.
At least this provides a case of a broadcasted event, which requires
per-event data allocation.
See github issue #576.
May or may not be useful in some ways.
We require a context parameter for this just to be sure, even if the
internal implementation currently doesn't.
That's one less mpv internal function for the Lua wrapper.
Also mention that NULL isn't valid. Although I'm not sure whether the
implementation strictly follows this (it should, but there are some
wacky corner cases).
There are some complications because the client API distinguishes
between integers and floats, while Lua has only "numbers" (which are
usually floats). But I think this should work now.
This is only half-implemented: actually the option will first be
converted from mpv_node to its native type, then it's converted to a
string, and then back to its native type. This is because the option
API was made for strings and not anything else.
Other than being grossly inelegant, the only downside is probably with
string lists and key/value lists, which don't escape strings containing
syntax elements correctly.
This automatically allows accessing properties like chapter-list and
track-list to be read as mpv_node. This affects all properties which use
m_property_read_sub() and m_property_read_list().
This actually makes use of the client.h declarations and the mpv_node
mechanisms added some commits ago.
For now, using MPV_FORMAT_STRING will usually fallback to explicit
string conversion, but not in the other cases. E.g. reading a numeric
property as string will work, but not reading a string property as
number. Other than that, only MPV_FORMAT_INT64->MPV_FORMAT_DOUBLE does
an automatic conversion.
I'm not sure whether these semantics and API are good, so comments and
suggestions are welcome.
Allows retrieving properties by their native values (or something close
to it), rather than having to go through string conversion. The caller
could actually just copy the value itself and then use the m_option
functions to convert it to mpv_node, but maybe it's more flexible this
way.
m_option is basically the mechanism to handle C data types in a dynamic
way. Add functions to convert values to and from mpv_node. For example,
string lists are turned into mpv_node using MPV_FORMAT_NODE_ARRAY, and
so on.
This adds declarations for new formats. The implementation will be added
in the following commits. (It still compiles and runs with this commit,
because it adds constants only.)
The obvious new types are MPV_FORMAT_FLAG, MPV_FORMAT_INT64,
MPV_FORMAT_DOUBLE. MPV_FORMAT_FLAG is a boolean, but to avoid nasty ABI
issues or with languages that don't have a bool data type (C89), it uses
int. Thus the format is not named MPV_FORMAT_BOOL, to avoid confusion.
The MPV_FORMAT_NONE type (mpv_node) is a generic structured type, like a
variant or, say, JSON. It can store strings/bools/numbers, as well as
arrays and key/value pairs (with string keys only).
The MPV_FORMAT_NODE_ARRAY and MPV_FORMAT_NODE_MAP types are used
internally by mpv_node only and can't be used with most of the other API
(like mpv_set_property()) directly.
With mpv_set_property(h, "property", MPV_FORMAT_STRING, ptr), ptr now
has to be of type char** instead of char*. This makes it more consistent
with mpv_get_property() and also non-pointer formats, which will be
introduced in the following commits. mpv_set_property() of course does
not change its interface (only its implementation is adjusted to keep
its interface).
This also affects mpv_set_option(), but again not
mpv_set_option_string().
This is allowed in C99 and C++11, but apparently not in C89 and C++98.
Make it conform to the older standards, since we want the client API
header to be highly portable.
This uses the value of 1.95 as an approximation for the exact gamma
curve, which replicates the behavior of popular video software including
anything in the Apple ecosystem, as per issue #534.
Until now, strings were the only allowed dynamically allocated argument
type in input commands. Extend it so that it works for any type. (The
string expansion in command.c is of course still string specific.)
If this function could return the input value (i.e. the == case was
correct), then macros like MP_GROW_ARRAY would have been incorrect. The
implementation was correct though, so there's no bug.
Some code accessed m_option.name to get the property name. (Maybe only
show_property_osd() had a significant use of it.) Remove that, and
remove setting names and dummy names as well.
The old code usually assumed that the name was set, and
show_property_osd() used it to get the proper name of deprecated
aliases.
The "vf" property was listed as "vf*". Not sure why that was done, but
it works without anyway.
Doesn't require other code to care about this, which will allow us to
simplify the property code.
Only "wildcard" options like "vf" and string lists used this, and
m_option_list_findb() (which is excused).
M_OPT_PARSE_ESCAPES was pretty stupid, and broke the (useful) assumption
that string variables contain exactly the same value as set by the
option. Simplify it, and move escape handling to the place where it's
used.
Escape handling itself is not terribly useful, but still allows useful
things like multiline custom OSD with "\n".
MP_CMD_COMMAND_LIST commands (used to implement key bindings with
multiple commands) were not checked for abort commands. Implement it.
Remove the remarks about multi-commands being special from the manpage.
Seek coalescing is handled differently now, and the issue with abort
commands is fixed with this commit.
Current code stolen from waf's extras, only supported 'pe' and 'elf'. OS X
uses the 'Mach-O' binary format (which waf calls 'mac-o'... go figure).
Add support for generating the global symbols file with nm and using it from
clang.
The old way still works, and is fine to use. Still discourage it,
because it might conflict with other ways to access this property, such
as the one added in the next commit.
This is a regression introduced from moving Lua scripts (including the
OSC) to their own threads. Now OSC and dvdnav can add their bindings at
the same time without coordination, which seems to result in the OSC
winning most time, and thus overriding the dvdnav menu bindings.
Fix this by adding a flag that makes dvdnav menu bindings take priority
over all other bindings.
When timeline was used, and the --start option was not used, the initial
seek (needed to switch to the first timeline segment) seeked to -1 due
to an oversight.
This is simply not important enough to warrant so much space, and it's
perhaps also very confusing.
Although I'm not fully sure, since this is about the only way that
allows a user to interact with a script, besides key bindings and static
options.
Detected 'protocols' are AFP, nfs, smb and webdav. This can be extended on
request.
This is currently only implemented for BSD systems (using fstatfs). This
addresses issue #558 on the above platforms.
There was already an undocumented mechanism provided by
mp.set_key_bindings and other functions, but this was relatively
verbose, and also weird. It was mainly to make the OSC happy (including
being efficient and supporting weird corner cases), while the new
functions try to be a bit simpler.
This also provides a way to let users rebind script-provided commands.
(This mechanism is less efficient, because it's O(n^2) for n added key
bindings, but it shouldn't matter.)
- Adds description of and uses $JOBS envvar in MXE instructions
- Adds MXE_TARGETS to command line instead of echoing it to settings.mk
- Prettify and sentence usage
This adds a mechanism for easier export of sub-properties. The following
commits will make use of it to export fine grained information about
certain things. The sub-property mechanism reduces the amount of code
needed to export a data value to 1 line.
* 'master' of git://github.com/mpv-player/mpv:
win32: restore support for exe directory as config directory
crosscompile-mingw: improve instructions for MXE
sd_lavc: handle subtitles with no subtitle resolution set
options: make --no-config block all auto-loaded configuration files
lua: auto-load scripts from ~/.mpv/lua/
lua: make register_event() not overwrite previous event handler
It seems that it was causing issues with certain perl setups (such as
the one on issue #549). It also turns out that it was not behaving correctly
(not all constants were being promoted to big nums as they should), so we
use explicit objects to derive the constants.
There were also precedence issues. I wonder if this even worked right to
begin with.
The 'double' path (8-byte floats) is untested, as I couldn't easily find
a file with such a field.
Closes#549.
Set subtitle resolution to video resolution when avctx->width and
avctx->height are zero.
This can happen with broken vobsubs that have no size set in their
.idx file (or Matroska extradata). At least with the test file provided
in issue #551, using the video resolution as fallback instead of what
guess_resolution() does is better.
Note that these files clearly are broken. It seems this particular
file was created by trying to use ffmpeg to transcode DVB subtitles
to vobsub, and ffmpeg "forgot" to set the subtitle resolution in the
destination file. On the other hand, ffmpeg DVB and PGS decoders set
the resolution on the first subtitle packet (or somewhere close), so
it's not really clear what to do here.
Closes#551.
Signed-off-by: wm4 <wm4@nowhere>
Patch by xylosper, rewritten commit message by wm4.
Until now, the --no-config was explicitly checked in multiple places to
suppress loading of config files.
Add such a check to the config path code itself, and refuse to resolve
_any_ configuration file locations if the option is set.
osc.lua needs a small fixup, because it didn't handle the situation when
no path was returned. There may some of such cases in the C code too,
but I didn't find any on a quick look.
Instead, chain them.
Note that there's no logic to prevent the other event handlers to be run
from an event handler (like it's popular in GUI toolkits), because I
think that's not very useful for this purpose.
Cygwin's libc (newlib) doesn't obey a lot of unix feature test macros,
including _GNU_SOURCE; as a result, a lot of functions and defines get
masked out -- important defines such as M_PI and strcasecmp. Work around
it by undefining __STRICT_ANSI__ on cygwin systems.
This will still cause compilation issues on any non-cygwin system that
uses newlib, but hopefully nobody does that, or if they do, they will
find this commit message and know to add -U__STRICT_ANSI__ to their
CFLAGS. Hopefully.
The mplayer decoder (spudec.c) actually handled this. There was explicit
code for binary palettes (16 32 bit values), and the subtitle resolution
was handled by video resolution coincidentally matching the subtitle
resolution.
Whoever puts vobsub into mp4 should be punished.
Fixes the sample gundam_sample.mp4, closes github issue #547.
In particular, this affects drag & drop of subtitles, which uses sub_add
internally. This will make the subtitles show up immediately, instead of
requiring manual selection of the added subtitle.
Might be not so ideal when adding multiple subtitles at once, because
that leads to multiple sub_add commands, and will end up with the last
subtitle instead of the first selected. But this is a minor detail.
The minimum required version was bumped in the old configure script, but
for the waf build system is was somehow forgotten or overlooked.
Probably happened while the waf build system was developed in a separate
branch.
Closes#546.
This is the same issue as addressed by 257d9f1, except this time for
the :srgb option as well. (257d9f1 only addressed :icc-profile)
The conditions of the srgb_compand mix() call are also flipped to
prevent an off-by-one error.
Use a list instead of a table. This makes it easier to provide extended
information about a property, and doesn't require you to fiddle with rhe
RST ASCII-art tables.
Also, extend some property descriptions.
The initialization code was split and refactored for the libmpv changes.
One change, moving a part of cocoa initialization, accidentally broke
--force-window on OSX, which creates a VO in a certain initialization
stage. We still don't know how cocoa should behave with libmpv, so fix
this with a hack to beat it back into working. Untested.
I was unhappy with the old way of handling buffers, especially resizing. But my
original plan to use wl_shm_pool_resize wasn't as good as I initially thought.
I might get back to it.
With the new buffer pools it now possible to select triple buffering. Also the
buffer pools are also needed for the upcoming subsurfaces for osd and subtitles.
I hope this change was worth it.
Use of these is "discouraged", but they're there to select these special
cases with the "aspect" property. They really should use some sort of
choice option type, but since it would be some work to make these work
with float values, the simple and dumb alternative was picked.
Return the error Lua-style, instead of raising it as Lua error. This is
better, because raising errors is reserved for more "fatal" conditions.
Pretending they're exceptions and trying to do exception-style error
handling will just lead to pain in this language.
Print a warning if a library has mismatched compile time and link time
versions.
Refuse to work if the compile time and link time versions are a mix of
ffmpeg and libav. We print an error message and call exit(). Since we'd
randomly crash anyway, I think this is ok.
This doesn't catch the case if you e.g. use a ffmpeg libavcodec and a
libav libavformat, which would of course just crash as quickly, but I
think this checks enough already.
This library will export the client API functions.
Note that this doesn't allow compiling the command line player to link
against this library yet. The reason is that there's lots of weird stuff
required to setup the execution environment (mostly Windows and OSX
specifics), as well as things which are out of scope of the client API
and every application has to do on its own. However, since the mpv
command line player basically reuses functions from the mpv core to
implement these things, it's not very easy to separate the command
line player form the mpv core.
This is partial only, and it still accesses some MPContext internals.
Specifically, chapter and track lists are still read directly, and OSD
access is special-cased too.
The OSC seems to work fine, except using the fast-forward/backward
buttons. These buttons behave differently, because the OSC code had
certain assumptions how often its update code is called.
The Lua interface changes slightly.
Note that this has the odd property that Lua script and video start
at the same time, asynchronously. If this becomes an issue, explicit
synchronization could be added.
Add a client API, which is intended to be a stable API to get some rough
control over the player. Basically, it reflects what can be done with
input.conf commands or the old slavemode. It will replace the old
slavemode (and enable the implementation of a new slave protocol).
This avoids trouble if another mpv instance is initialized in the same
process.
Since timeBeginPeriod/timeEndPeriod are hereby not easily matched
anymore, use an atexit() handler to call timeEndPeriod, so that we
can be sure these calls are matched, even if we allow multiple
initializations later when introducing the client API.
It's quite possible to overflow the calculation by setting the timeout
to high values. Limit it to INT_MAX, which should be safe. The issue is
mainly the secs variable.
timespec.tv_sec will normally be 64 bit on sane systems, and we assume
it can't overflow by adding INT_MAX to it.
This skipped all audio packets before the first video key frame was
found. I'm not really sure why this would be needed; most likely it
isn't. So get rid of it. Even if audio packets are returned to the
player too soon, the player will sync the audio start to the video
start by decoding and discarding audio data.
Note that although the removed code was just added in the previous
commit, it merely kept the old keeping semantics which demux_mkv
always followed. This commit removes these special semantics.
v_skip_to_keyframe is set to true while non-keyframe video packets are
skipped. Until now, audio packets were also skipped when doing this. I
can't see any good reason why this would be done, but for now I want to
keep the old logic when audio+video seeks are done.
However, for audio-only mode, do proper seeking, which also fixes
behavior when trying to seek past the end of the file: playback is
terminated properly, instead of starting playback on the start of the
last cluster.
Note that a_no_timecode_check is used only for audio+video seek. I'm
not sure what this is needed for, but it might influence A/V sync after
seeking.
This sometimes happened when changing playback speed (= reinitializing
audio) after seeking of playback start. The assertion in audio.c:441 was
triggered, because buffer_playable_samples wasn't reset correctly when
the audio buffer was cleared or shortened. The assertion is correct and
should hold up any time.
I could not see any difference whatsoever, but for usage with a 3DLUT
there's zero performance difference so we might as well follow the spec to
the letter.
On Windows, no ANSI control sequences are available, so we can't easily
clear lines, move the cursor, etc. It's yet to be decided how this
should be handled (emulate ANSI escapes in osdep/terminal-win.c, or
provide abstracted terminal API functions to unify the Linux and Windows
code).
For now, this fixes the regression that was introduced earlier by the
status line rewrite. It doesn't fix all aspects of status line and
terminal OSD handling, as can be clearly seen by the unconditional use
of terminal_erase_to_end_of_line further down the changed code.
Fixes github issue #499 (sort of).
Trying to set a non-existent flag (like +keepside on Libav) causes
libavutil print an incomprehensible warning (something about eval;
probably the overengineered libavutil option parser tripping over the
'+' normally used for flags, and trying to interpret it as formula).
There's apparently no easy way to check for the existence of a flag,
so add some more ifdeffery to shut it up.
The code removed from handle_input_and_seek_coalesce() did two things:
1. If there's a queued seek, stop accepting non-seek commands, and delay
them to the next playloop iteration.
2. If a seek is executing (i.e. the seek was unqueued, and now it's
trying to decode and display the first video frame), stop accepting
seek commands (and in fact all commands that were queued after the
first seek command). This logic is disabled if seeking started longer
than 300ms ago. (To avoid starvation.)
I'm not sure why 1. would be needed. It's still possible that a command
immediately executed after a seek command sees a "seeking in progress"
state, because it affects queued seeks only, and not seeks in progress.
Drop this code, since it can easily lead to input starvation, and I'm
not aware of any disadvantages.
The logic in 2. is good to make seeking behave much better, as it
guarantees that the video display is updated frequently. Keep the core
idea, but implement it differently. Now this logic is applied to seeks
only. Commands after the seek can execute freely, and like with 1., I
don't see a reason why they couldn't. However, in some cases, seeks are
supposed to be executed instantly, so queue_seek() needs an additional
parameter to signal the need for immediate update.
One nice thing is that commands like sub_seek automatically profit from
the seek delay logic. On the other hand, hitting chapter seek multiple
times still does not update the video on chapter boundaries (as it
should be).
Note that the main goal of this commit is actually simplification of the
input processing logic and to allow all commands to be executed
immediately.
Legacy GL context creation (glXCreateContext) explicitly requires a X
visual, while the modern one (glXCreateContextAttribsARB) does not for
some reason. So fail only on the legacy code path if we don't find a
visual. Note that vo_x11_config_vo_window() will select a default visual
if a NULL visual is passed to it.
This fixes issue #504. For some reason, glXChooseFBConfig() will return
a fbconfig with no associated visual. (I'm not sure if this allowed.
They don't always have a visual, but since GLX_X_RENDERABLE is set
and GLX_DRAWABLE_TYPE is (implicitly) set to GLX_WINDOW_BIT, why would
there be no visual?)
Even worse, a test program seems to show that a 16 bit fbconfig is
selected (instead of 24/32 bit), which doesn't sound nice at all. Since
there _are_ better fbconfigs available, glXChooseFBConfig() should
normally sort them by quality, and return the better ones first. It's
worth noting that this function should also prefer GLX_TRUE_COLOR
over anything else, although this comes last in the sort order.
Whatever is going on, requesting GLX_X_VISUAL_TYPE with GLX_TRUE_COLOR
seems to fix it.
There is some logic to discard packets from streams that are not
selected. Run the metadata update code before this, just to make 100%
sure that no metadata updates can be lost when streams are deselected.
(I'm not sure why this logic would be needed, since both libavformat and
the generic demuxer code do this already. But a quick test shows that
av_read_frame() can return a packet from a stream even if the stream has
AVStream.discard set to AVDISCARD_ALL. This happened after stream
switching. Maybe libavformat doesn't discard already queued packets.)
Instead of printing lines like:
Demuxer info GENRE changed to Alternative Rock
Just output all tags once they change. The assumption is that individual
tags rarely change, while all tags change in the common case.
This changes tag updates to use polling. This could be fixed later,
although the ICY stuff makes it a bit painful, so maybe it will remain
this way.
Also remove DEMUXER_CTRL_UPDATE_INFO. This was intended to check for tag
updates, but now we use a different approach.
Use an arbitrary constant instead, which is as good as PATH_MAX.
This helps us to avoid having to think about pull request #523.
Also fix a case where a potentially signed char was passed to isspace().
This was done incorrectly in the previous commit: the fallback size used
the window size as requested with the first config call, which is the
size of the hidden window in the vo_opengl case. (That damn hidden
window again...)
This code essentially does nothing. As far as I could find out, this
actually used to do something. Then it was removed with commit efe7c39f,
leaving some leftover code that didn't do anything useful. This happened
12 years ago!
Also remove a commented debug printf.
vo_opengl creates a hidden X11 window to probe the OpenGL context. It
must do that before creating a visible window, because VO creation and
VO config are separate phases.
There's a race condition involving the hidden window: when starting with
--fs, and then leaving fullscreen, the unfullscreened window is
sometimes set to the aspect ratio of the hidden window. I'm not sure why
the window size itself uses the correct size (but corrupted by the wrong
aspect), but that's perhaps because the window manager is free to ignore
the size hint while honoring the aspect, or something equally messed up.
It turns out this happens because x11_common.c thinks the size of the
hidden window is the size of the unfullscreened window. This in turn
happens because vo_x11_update_geometry() reads the size of the hidden
window when called in vo_x11_fullscreen() (called from
vo_x11_config_vo_window()) when mapping the fullscreen window. At that
point, the window could be mapped, but not necessarily. If it's not
mapped, it will get the size of the unfullscreened window... I think.
One could fix this by actively waiting until the window is mapped. Try
to pick a less hacky approach instead, and never read the window size
until MapNotify is received.
vo_x11_create_window() needs a hack, because we'd possibly set the VO's
size to 0, resulting e.g. in vdpau to fail initialization. (It'll print
error messages until a proper resize is received.)
This fixes a weird bug with aspect ratio handling. It has to do with
float handling: with -std=gnu99, gcc implicitly enables broken non-
standard semantics giving float variables excess precision. This can for
example make this fail in theory: "float a = 0.1; assert(a == a);"
While standard C allows excess precision _within_ expressions, it
requires truncation when storing float values in variables of types
"float" or "double". The "gnu99" mode breaks this. It can be unbroken by
using "c99", or by specifying -fexcess-precision=standard. The former
seems less likely to break compilers other than modern gcc. Note that
-ffloat-store would also fix this, but also makes float expressions less
efficient and less precise for no reason.
The code that mistakenly fails because of this is dec_video.c line 393.
It caused the container aspect to be ignored in some or all situations,
depending how the compiler optimizes. For example, on gcc-4.6 with -Os,
the aspect is always ignored.
In future, we should probably just get rid of storing aspects as floats.
Note that this still happens in the stream level, so we can't have
nice highlevel behavior restricting seeking. Instead, if a seek leads
to the demuxer requesting data outside of the cached range, the seek
will simply fail. This might confuse the demuxer, and the resulting
behavior is not necessarily useful.
Note that this also doesn't try to skip data on a forward seek. This
would just freeze the stream with slow unseekable streams.
One nice thing is that stream.h has a separate function for merely
skipping data (separate from seeking forward), which is pretty useful
in this case: we want skipping of data to work, even if we reject
seeking forward by skipping data as too expensive. This probably is
or will be useful for demux_mkv.c.
Usually, you have to call pthread_cond_timedwait() in a loop (because it
can wake up sporadically). If this function is used by another higher
level function, which uses a relative timeout, we actually have to
reduce the timeout on each iteration - or, simpler, compute the
"deadline" at the beginning of the function, and always pass the same
absolute time to the waiting function.
Might be unsafe if the system time is changed. On the other hand, this
is a fundamental race condition with these APIs.
This used to work; I'm not sure when or why it regressed. When setting
AVProbeData.filename to NULL, libavformat will crash in rtp_probe() by
unconditionally accessing the string.
We used to set the filename to NULL to prevent probing by file extension
when we don't deem it as necessary. Using an empty string also works for
this purpose.
Try to make it more intuitive by not requiring hex values. The new way
uses float values in the range 0.0-1.0, separated by '/' (':' was
suggested, but that wouldn't allow color options in sub-options).
Example: --osd-color=1.0/0.0/0.0/0.75
Using the range 0.0-1.0 has the advantage that it could be easily
extended to colors beyond 8 bit.
Details see manpage.
Suggestions for alternative syntax or value ranges are welcome, but be
quick with it.
This avoids stray newlines when:
1. Some (non-status line) text was output
2. Then an empty status line is output
According to the logic, 2. should print an empty line to show the blank
status line. Don't do that, and instead output nothing in this case.
This caused problems with mpv_identify.sh, and also looked ugly when
using --quiet.
Larger sizes can introduce overflows, depending on the image format. In
the worst case, something larger than 16000x16000 with 8 bytes per pixel
will overflow 31 bits.
Maybe there should be a proper failure path instead of a hard crash, but
not yet. I imagine anything that sets a higher image size than a known
working size should be forced to call a function to check the size (much
like in ffmpeg/libavutil).
Not everything in the OSD path handles 0x0 sized sub-bitmaps well. At
least the code implementing --sub-gray had severe problems with it.
Fix this by skipping such bitmaps.
RGB565 is one of the fastest and most supported formats on low end consumer
devices, but ffmpeg spams warning when using it. Make it opt-in instead of
opt-out.
The problem seems to have solved itself. I guess the previous changes to
resizing and commit ba101ab made this possible. Consider me happy for removing
that crap.
Windows applications that use LoadLibrary are vulnerable to DLL
preloading attacks if a malicious DLL with the same name as a system DLL
is placed in the current directory. mpv had some code to avoid this in
ao_wasapi.c. This commit just moves it to main.c, since there's no
reason it can't be used process-wide.
This change can affect how plugins are loaded in AviSynth, but it
shouldn't be a problem since MPC-HC also does this and it's a very
popular AviSynth client.
Windows users expect this when a program crashes. Without it, the
program just disappears. Also change the SetErrorMode call to use macros
instead of a hardcoded constant.
If, for some reason, the subtitle renderer attempts to render a
subtitle before SD_CTRL_SET_VIDEO_PARAMS was called, it passed a
value calculated from invalid values. This can happen with --vf=sub
and --start. The crash happens if 1. there was a subtitle packet that
falls into the timestamp of the rendered video frame, 2. the playloop
hasn't informed the subtitle decoder about the video resolution yet
(normally unneeded, because that is used for weird corner cases only,
so this code is a bit fuzzy), and 3. something actually requests a
frame to be drawn from the subtitle renderer, like with vf_sub.
The actual crash was due to passing NaN as pixel aspect to libass,
which then created glyphs with ridiculous sizes, involving a few
integer overflows and unchecked mallocs.
The sd_lavc.c and sd_spu.c cases probably don't crash, but I'm not
sure, and it's better fix them anyway.
Not bothering with sd_spu.c, this crap is for compatibility and will
be removed soon.
Note that this would have been no problem, had the code checked whether
SD_CTRL_SET_VIDEO_PARAMS was actually called. This commit adds such a
check (although it basically checks after using the parameters).
Regression since 49caa0a7 and 633fde4a.
This generally affects mp3 files that don't have any (or many) mp3
frames in the first 2 MB. 2 MB is the maximum probe size, and
libavformat returns a low probescore even if we give it the full 2 MB.
Trying to probe a larger buffer (or even the full file) doesn't work for
mysterious reasons.
The workaround consists in accepting a very weak probescore if the
format is detected as mp3 and we probed already 2 MB.
Restructure it a bit, so we can use the format hack list even if no mime
type applies. Shouldn't change anything functionally yet. Preparation
for the next commit.
Still untested, because now it crashes inside of libSDL for unknown
reasons. (This also happens with mpv git from yesterday - probably an
installation problem, or SDL doing weird things it shouldn't be doing.)
The main difference between the old and new callbacks is that the old
callbacks required passing the window size, which is and always was very
inconvenient and confusing, since the window size is already in
vo->dwidth and vo->dheight.
Balance controls as used by mixer.c was broken, because af_pan.c stopped
accepting its arguments. We have to allow 0 channels explicitly. Also,
fix null pointer access if the matrix parameter is not used.
Regression from commit 82983970.
If there's more than one edition, print the list of editions, including
the edition name, whether the edition is selected, whether the edition
is default, and the command line option to select the edition. (Similar
to stream list.)
Move reading the tags to a separate function process_tags(), which is
called when all other state is parsed. Otherwise, that tags will be lost
if chapters are read after the tags.
Pretty worthless. This is called from the seek code, which will
reinitialize these anyway. Even if seeking somehow decides to fail, the
new values are still valid.
One could say a failed seek (if that happens) should jump back to the
original position, and thus it would be better to make sure the state
is restored. But then demux_mkv_seek needs to do this correctly,
including not setting up skipping to the target timestamp. But not
bothering with this.
Extremely obscure corner case with concatenated segments, in which EOF
wasn't recognized correctly, and it tried to demux clusters from the
next segment.
See [MKV]_Editions,_Linked_Segments,_&_Tracksets.mkv from the CCCP test
file collection.
This basically used to be part of the user interface, before mpv moved
printing the track list to the frontend, and this code was raised to
verbose output level.
For some reason, if an error happened when reading headers, it merely
stopped reading the headers, and then continued normally. (It looks like
the case to exit hard (-2) was mainly used for skipping unwanted ordered
chapter segments.)
I can't comprehend this. Always exit on error when reading headers.
(Maybe some more error tolerance would be good, but I have no test case,
and there's some danger of entering endless loops.)
This makes everything more robust, and also somewhat simpler (even if
the diffstat isn't very impressive).
Instead of recursively following SeekHeads while reading headers, just
read the headers until the first cluster, and then possibly use
SeekHeads to read the remaining missing headers.
Rename vo_get_src_dst_rects() to mp_get_src_dst_rects() and make it
independent from the VO (it takes a comical amount of parameters now to
pass all required state). Add a convenience wrapper with the name
vo_get_src_dst_rects() to vo.c. Replace all aspdat and vo usages with
immediate parameters.
Functionally, nothing should change, except that the window size is
clamped to a minimum of size 1 much earlier, and some log messages
change the prefix (don't bother with vo.vo_log stuff).
The plan is to make all the code in aspect.c independent from vo.c,
which should make the code easier to understand, will allow removal of
vo->aspdat, and reduces the amount of code that accesses weird mutable
struct vo fields.
vo->aspdat is basically an outdated version of vo->params, plus some
weirdness. Get rid of it, which will allow further cleanups and which
will make multithreading easier (less state to care about).
Also, simplify some VO code by using mp_image_set_attributes() instead
of caring about display size, colorspace, etc. manually. Add the
function osd_res_from_image_params(), which is often needed in the case
OSD renders into an image.
Some mpv builds identify with e.g. "mpv b'0.3.3' ". The version looks
like str() was called on a Python byte string. I couldn't reproduce it
on my machine (I tried with both Python 2 and 3), so I'm not exactly
sure what's going on here, but I'm hoping this commit does fix it.
This was broken yesterday: the playlist demuxer will always fall back to
plaintext playlist files, which will cause the ASX playlist parser and
some others never to be called.
This is probably ok. Probing could hit this case very often, since it'll
mean running this function on potentially binary data, but on the other
hand, probing usually uses a memory stream (to limit the amount of data
read), and memory streams have s->log silenced (details see
open_memory_stream()).
This simplifies the implementation and should make it more robust. For
example, we return an error if a line is longer than the provided buffer
(instead of splitting the line).
The code is much shorter, because now finding the new line and reading
characters is done in one go.
As of this commit, stream_read_line() can't actually error (except in
the case the passed in buffer is 0, which never happens here). This
commit is preparation for the following commit, which checks harder
whether the read data is actually text. Before this commit, an error
was treated as end-of-file, but the data read so far was considered
valid.
Signed-off-by: wm4 <wm4@nowhere>
This merges pull request #496. The problem was that at least the
initialization of the distance[] array accessed af_fmtstr_table[]
entries that were out of bounds. Small cosmetic changes applied to
the original pull request.
Doesn't affect the generated code, but avoids confusion
in both humans and newer Coverity versions.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@36623 b3059339-0415-0410-9bf9-f77b7e298cf2
The struct we need to copy is actually a cdrom_msf0, not cdrom_msf.
Even though the kernel for no good reason reads it in as a
cdrom_msf struct, but only uses the part shared with cdrom_msf0 -
this is probably a kernel bug.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@36622 b3059339-0415-0410-9bf9-f77b7e298cf2
Do two things:
1. add locking to struct osd_state
2. make struct osd_state opaque
While 1. is somewhat simple, 2. is quite horrible. Lots of code accesses
lots of osd_state (and osd_object) members. To make sure everything is
accessed synchronously, I prefer making osd_state opaque, even if it
means adding pretty dumb accessors.
All of this is meant to allow running VO in their own threads.
Eventually, VOs will request OSD on their own, which means osd_state
will be accessed from foreign threads.
The plan is to make the whole OSD thread-safe, and we start with this.
We just put locks on all entry points (fortunately, dec_sub.c and all
sd_*.c decoders are very closed off, and only the entry points in
dec_sub.h let you access it). I think this is pretty ugly, but at least
it's very simple.
There's a special case with sub_get_bitmaps(): this function returns
pointers to decoder data (specifically, libass images). There's no way
to synchronize this internally, so expose sub_lock/sub_unlock functions.
To make things simpler, and especially because the lock is sort-of
exposed to the outside world, make the locks recursive. Although the
only case where this is actually needed (although trivial) is
sub_set_extradata().
One corner case are ASS subtitles: for some reason, we keep a single
ASS_Renderer instance for subtitles around (probably to avoid rescanning
fonts with ordered chapters), and this ASS_Renderer instance is not
synchronized. Also, demux_libass.c loads ASS_Track objects, which are
directly passed to sd_ass.c. These things are not synchronized (and
would be hard to synchronize), and basically we're out of luck. But I
think for now, accesses happen reasonably serialized, so there is no
actual problem yet, even if we start to access OSD from other threads.
These were needed before the last commit, but now they don't do anything
anymore. (They were used to decide whether to replace or stack the
previous OSD message when a new one was displayed.)
If certain OSD messages were displayed at the same time, the hidden
messages were put on the stack, and displayed again once the higher
priority messages disappeared. The idea was probably that lower priority
messages could not hide higher priority ones, and also that the lower
messages did not get lost.
But in practice, this gives confusing results with OSD messages randomly
reappearing for a brief time. Remove it.
Showing subtitles on terminal used the OSD message stack (which uses a
stack to "pile up" messages that were displayed at the same time). This
had a bunch of weird and annoying consequences. This accessed a certain
osd_state field, which is a minor annoyance since I want to make that
struct opaque. Implement this differently.
The values set by this new option can be queried by Lua scripts using
the mp.getopt() function. The function takes a string parameter, and
returns the value of the first key that matches. If no key matches, nil
is returned.
When the Lua code was written, the core didn't have names for log levels
yet (just numbers). The only user visible change is that "verbose"
becomes "v", since this level had different names.
Adds the following Lua function to enable message events:
mp.enable_messages(size, level)
size is the maximum number of messages the ringbuffer consists of. level
is the minimum log level for a message to be added to the ringbuffer,
and uses the same values as the mp.log() function. (Actually not yet,
but this will be fixed in the following commit.)
The messages will be delivered via the mp_event() in the user script,
using "message" as event name. The event argument is a table with the
following fields:
level: log level of the message (string as in mp.log())
prefix: string identifying the module of origin
text: contents of the message
As of currently, the message text will contain newline characters. A
message can consist of several lines. It is also possible that a
message doesn't end with a newline, and a caller can use multiple
messages to "build" a line. Most messages will contain exactly 1 line
ending with a single newline character, though.
If the message buffer overflows (messages are not read quickly enough),
new messages are lost until the queued up messages are read. At the
point of the overflow, a special overflow message is inserted. It will
have prefix set to "overflow", and the message text is set to "".
Care should be taken not to print any messages from the message event
handler. This would lead to an infinite loop (the event handler would be
called again after returning, because a new message is available). This
includes mp.log() and all mp.msg.* functions. Keep in mind that the Lua
print() function is mapped to mp.msg.info().
Until now, mp_msg output always went to the terminal. There was no way
to grab the stream of output messages. But this will be needed by
various future changes: Lua scripts, slave mode, client library...
This commit allows registering a ring buffer. A callback would be more
straight-forward, but since msg.c sits at the bottom of the lock
hierarchy (it's used by virtually everything), this would probably be a
nightmare. A ring buffer will be simpler and more predictable in the
long run.
We allocate new memory for each ringbuffer entry, which is probably a
bit expensive. We could try to be clever and somehow pack the data
directly into the buffer, but I felt like this wouldn't be worth the
complexity. You'd have to copy the data a bunch of times anyway. I'm
hoping that we can get away with using the ringbuffer mechanism for
low frequency important messages only (and not e.g. for high volume
debug messages), so the cost doesn't matter that much.
A ringbuffer has a simple, single log level. I considered allowing
--msglevel style per-prefix configuration for each ringbuffer, but
that would have been pretty complicated to implement, and wouldn't
have been that useful either.
This makes
mp_msg(x, y, "a\nb\n")
behave the same as
mp_msg(x, y, "a\n")
mp_msg(x, y, "b\n")
which is probably what one would expect. Before this commit, the "b"
line didn't have a prefix when using ths single mp_msg call.
Before that, it just returned -1.
The print case is inconsistent with that, but I'll leave it for now,
because it's consistent with status line / show_progress behavior.
This is a bit of a hack, but in order to prevent TranslateMessage from
seeing WM_KEYDOWN messages that we already know how to decode, move the
decoding logic to the event loop. This should fix#476, since it stops
the generation of extraneous WM_CHAR messages that were triggering more
than one action on keydown.
Starting a network stream could stall by executing uncacheable stream
control requests (STREAM_CTRL_GET_LANG and STREAM_CTRL_GET_DVD_INFO).
Being uncacheable means the player has to wait until the cache is done
reading the current block of data. These requests can't be cached
because they're too complicated, so the only way to avoid them is
special casing the DVD and Bluray streams (which are the only things
which need these requests), and not doing them in other cases.
(This is kind of inelegant, but so is the rest of the DVD/BD code.)
Application icon was added to the Dock only when run inside of a bundle. That
was handled automatically by OS X using the Info.plist definition.
To add the Application icon when run as a CLI program, I used the samme
approach in the X11 code and loaded the icon as a static binary blob inside
of mpv's binary. This is the simplest approach as it avoid headackes when
relocating the binary and such.
Many ebml_read_* functions have a length int pointer parameter, which
returns the number of bytes skipped. Nothing actually needed this
(anymore), and code using it was rather hard to understand, so get rid
of them.
Matroska makes it pretty hard to resync correctly on broken files:
random data returns "valid" EBML IDs with a high probability, and when
trying to skip them it's likely that you skip a random amount of data
(instead of considering the element length invalid).
Improve upon this by skipping known level 1 elements only. Consider
everything else invalid and call the resync code. This might result in
annoying behavior when Matroska adds new level 1 elements, although it
won't be particularly harmful. Matroska doesn't really allow us to do
better (even mkvtoolnix explicitly checks for known level 1 elements).
Since we now don't always want to combine EBML element skipping and
resyncing, remove ebml_read_skip_or_resync_cluster(), and make
ebml_read_skip() more tolerant against skipping broken elements.
Also, don't resync when reading sub-elements, and instead do resyncing
when reading them results in an error.
Until now, corrupted files were detected if the size of an element (that
should be skipped) was larger than the remaining file. This still could
skip larger regions of the file itself if the broken size happened to be
within the file.
Change it so that it's never allowed to skip outside the parent's
element.
The terminal OSD code includes the handling of the terminal status line,
showing player OSD messages on the terminal, and showing subtitles on
terminal (the latter two only if there is no video window, or if
terminal OSD is forced).
This didn't handle some corner cases correctly. For example, showing an
OSD message on the terminal always cleared the previous line, even if
the line was an important message (or even just the command prompt, if
most other messages were silenced).
Attempt to handle this correctly by keeping track of how many lines the
terminal OSD currently consists of. Since there could be race conditions
with other messages being printed, implement this in msg.c. Now msg.c
expects that MSGL_STATUS messages rewrite the status line, so the caller
is forced to use a single mp_msg() call to set the status line.
Instead of littering print_status() all over the place, update the
status only once per playloop iteration in update_osd_msg(). In audio-
only mode, the status line might now be a little bit off, but it's
perhaps ok.
Print the status line only if it has changed, or if another message was
printed. This might help with extremely slow terminals, although in
audio+video mode, it'll still be updated very often (A-V sync display
changes on every frame).
Instead of hardcoding the terminal sequences, use
terminfo/termcap to get the sequences. Remove the --term-osd-esc option,
which allowed to override the hardcoded escapes - it's useless now.
The fallback for terminals with no escape sequences for moving the
cursor and clearing a line is removed. This somewhat breaks status line
display on these terminals, including the MS Windows console: instead of
querying the terminal size and clearing the line manually by padding the
output with spaces, the line is simply not cleared. I don't expect this
to be a problem on UNIX, and on MS Windows we could emulate escape
sequences. Note that terminal OSD (other than the status line) was
broken anyway on these terminals.
In osd.c, the function get_term_width() is not used anymore, so remove
it. To remind us that the MS Windows console apparently adds a line
break when writint the last column, adjust screen_width in terminal-
win.c accordingly.
Seeking usually show the status on OSD. In terminal OSD mode, no status
is shown, because there is already a separate status line.
Unfortunately, the mechanism for showing the status was still active,
which forced showing no message while the code for showing seek status
was active.
Insane .ass subtitle scripts can cause severe slowdown (depending on the
speed of the machine, or the insanity of the script), so mention how to
test without subtitles. This is mainly to make the user aware that
subtitle rendering can be a problem. For longwinded explanation, there
isn't enough space.
Doesn't make any sense anymore. X11 (which was mentioned in the manpage)
autodetects it, and everything else ignored the option values.
Since for incomprehensible reasons the backends and vo.c still need to
exchange information about the screensize using the option fields,
they're not removed yet.
For some reason, this made all VO backends both set the screen
resolution in opts->screenwidth/height, and call
aspect_save_screenres(). Remove the latter. Move the code to calculate
the PAR-corrected window size from aspect.c to vo.c, and make it so that
the monitor PAR is recalculated when it makes sense.
When using --monitoraspect, but either the screen width or height or
both are unknown, a fallback is applied. This is a completely useless
obscure corner case that's going to help nobody, so get it out of the
way.
For a long time the cocoa backend set the xinerama_x/y and used dx/dy from the
VO instance. This somewhat worked with some workarounds but wasn't really
what was supposed to be happening. Moreover 27e4360, which touched this
workaround introduced a regression.
New code doesn't set the xinerama_x/y values so that dx/dy are offsets in the
current screen (not a virtual screen composed of all the screens). The screen
reference detected during VOCTRL_UPDATE_SCREENINFO is also passed down to the
window initialization code.
Fixes#472
Like with the previous commit, this is probably not needed, but it's
unclear whether that really is the case. Most likely, it used to be
needed by some demuxer, and now the only demuxer left that could
_possibly_ trigger this is demux_mkv.c.
Note that mjpeg is the only decoder that reads the extra_huff option,
and nothing in libavformat actually sets the option. So maybe it's
fundamentally not needed anymore.
This case can't happen with the normal realvideo codepath in
demux_mkv.c, because the code would errors out if the extradata is too
small, and everything would be broken anyway in the case the vd_lavc.c
condition is actually triggered.
It still might happen with VfW-muxed realvideo in Matroska, though.
Basically, I'm hoping this doesn't matter anyway, and that the vd_lavc.c
code was for other old demuxers, like demux_avi or demux_rm. Following
the commit history, it's not really clear for what demuxer this code
was added.
MicroDVD files _can_ contain real timestamps instead of frame timestamps
if they declare a FPS. But this seems to be rare, so ignore that if the
FPS happens to match with the libavformat microdvd parser's default FPS.
This might actually break files that declare 23.976 FPS, but the video
file is not 23.976 FPS, but the chance that this happens is probably
very low, and the commit fixes the more common breakage with 25 FPS
video.
Subtitle formats with frame based timing require using the video FPS to
compute proper subtitle timestamps. But it looks like the calculation to
do that was inversed.
This basically reverts the default as set by commit 812798c5. This seems
to be a matter of taste, but personally I think keeping the pause
setting is better.
Use the video chain for this instead. This is for facilitating coming
changes, which will clean up the vo->aspdat stuff, and this code would
be in the way.
On X11, if no wayland compositor is running, wl_list_init() will never
be called. This will cause destroy_display() to segfault when trying to
iterate over the list.
This fixes two things:
1. Dropping files on the VO window will auto-load subtitles (since most
drag & drop code prefixes the filenames with 'file://', and the
subtitle auto-load code considers 'file://' non-local)
2. Fix behavior of the %x screenshot filename template (similar problem)
One could force all that code to special-case 'file://' URLs, but just
replacing the filename on playback start is simpler.
The user_data is passed on add_listener and can later be changed with
set_user_data. But because we don't want to change it later and because it is
the same object remove the set_user_data call.
This might be a copy&paste leftover from the initial draft for the wayland
backend.
This finally gets rid of the LaTeX dependency.
We should actually be using docultils directly here, but I didn't
do this because of all the potential Python 2/3 breakage.
1000ms is a bit insane. It makes behavior on playback speed changes
worse (because the player has to catch up the dropped audio due to
audio-chain reset), and perhaps makes seeking slower.
Note that the problem of playback speed changes misbehaving will be
fixed in the future, but even then we don't want to have a buffer that
large.
I added enough logic to never set ontop or fullscreen twitce.
This commit keeps also the size of the video if multiple videos are played.
If the ratio differs the width will be kept at the same size and only the
height changes.
libwayland-client contains the following code [1]:
runtime_dir = getenv("XDG_RUNTIME_DIR");
if (!runtime_dir) {
fprintf(stderr,
"error: XDG_RUNTIME_DIR not set in the environment.\n");
This means this message will unconditionally and unavoidably be printed
if XDG_RUNTIME_DIR is not set. Since mpv is a terminal program, and we
want to avoid unnecessary output, work it around by not attempting to
use wayland if this environment variable is not set.
[1] http://cgit.freedesktop.org/wayland/wayland/tree/src/wayland-client.c#n636
(cd0dccd01e16fa404e03974d30ded3aebdb1c4bc)
This commonly happens when initializing vo_opengl on a X11-only system.
Unfortunately, most wl_*_destroy() functions appear not to accept NULL
pointers, making partial deinitialization a pain: you have to add your
own NULL checks everywhere to avoid crashes.
xkb.context is uninitialized separately, because you can initialize it
just fine, even if the rest of input initialization fails.
When both libavresample and libswresample were detected, the script
enabled both at the same time. This is not supported; although nothing
bad happened apparently. Make the dependencies both mutually exclusive.
This was inconsistent: the actual statusline used [statusline] as
message prefix, while other parts of the terminal OSD used [cplayer]
(and MSGL_STATUS). This commit makes it consistent.
Note that we can't use mp_msg, because it's not async-signal safe (we
might be running other threads while forking, so only functions
specified to be async-signal safe can be called, and this doesn't
include stdio; mp_msg acquires a mutex too).
Also, always print a \n before running the program to flush the status
line. The effect is that a program running successfully as well as the
error message will effectively start on a new line.
Quvi subtitles are considered external subtitles (simply because they're
separate from the audio/video stream), but for the sake of subtitle
auto-selection, they should not be considered external.
Change this so that quvi subtitles are treated like muxed subtitles
(with default flag never set). This means subtitles won't be selected by
default, unless explicitly requested with --sid or --slang.
Because of this commit there were problems displaying the frmase in their right
order.
This reverts commit 96e75d234a.
Conflicts:
video/out/gl_wayland.c
video/out/wayland_common.h
The changes in the vo_wayland_ontop function have no effect on the workaround.
Somehow the problem just disappeared. I guess it is because of the new control
function in gl_wayland.c where the resize happens immediatly after the event
dispatch/flush.
I don't like this function at all, but it's basically a trick to get the
input's mp_log instance in a case the mp_input_parse_cmd_strv() is
almost certainly not going to output anything. But still make it
somewhat more consistent with mp_input_parse_cmd_strv() - why force the
caller to always use MP_ON_OSD_AUTO?
Both X11 and Wayland support the same format for drag & drop operations
(text/uri-list), and the code for that was copied from x11_common.c to
wayland_common.c. Factor it out.
demux_subreader.c contains the old MPlayer subtitle parser, and I have
absolutely no confidence in this (very crappy) code. There might be
one or two security risks associated with running that code on
arbitrary input.
This solves the issue where we would not receive any frame events. The
difference to my earlier tests is that now it looks like eglSwapBuffers uses
it's own event queue or something similiar along the lines. Becaues the
performance is the same as without any redraw callback.
At the moment there are visual glitches when we resize the window. This happens
because in wayland there a special function for resizing EGL windows.
To prevent the glitches move the egl_context to the wayland state in
wayland_common.h and add a new control function to gl_wayland.c to wrap the
vo_wayland_control function to check for resize events.
With the new control wrapper the glitches are gone and the resizing is fluid.
The reason a segmentation happend here was because we couldn't get the
requested minor version. The major version is enough for differentiating
between OpenGL 3 and OpenGL 2. If it fails there is still a fallback to any
version available.
Also add a warning if we use the fallback.
Note that we don't try to be clever about detecting the files as
subtitles: we just check the file extension. We could go all the way and
check the files by opening them with a demuxer, but that would probably
do more bad than good.
Drag and drop is pretty complicated (just note how the number of Atoms
in use almost doubles), so I'm not sure whether this works everywhere.
This has been written by looking at the specification [1] or what looks
like the specification, and some external example code [2]. (The latter
one has no code license, but we didn't copy any code.)
We completely ignore the "requirement" of the spec. that the filename
"must" include username and hostname, e.g. "file://user@host/path/file".
In theory, this is required because X is network transparent, but at
this point the so called network transparency is a complete joke, and
Konqueror for one didn't include hostnames in "file://" URIs.
Tested with konqueror as drop source.
[1] http://www.newplanetsoftware.com/xdnd/
[2] http://www.edwardrosten.com/code/dist/x_clipboard-1.1/paste.cc
This is necessary to start mpv without forcing a console window,
but also breaks console usability. A workaround is to call mpv
from a wrapper process that uses the console subsystem and helps
redirecting the standard streams and WriteConsole output to where
they belong.
Streams like CDDA have special requirements in what quantities data can
be read: you can only read a sector at once, not more and not less. The
stream_peek() function didn't respect that and set less (used internal
buffer size of 2048 bytes, instead of CD sector size of 2352 bytes), so
no data was read and EOF was accidentally set, making playback with
cdda:// fail.
This is a regression since commit 9a723f, but that commit merely exposed
the issue (the redundant seek would clear the EOF flag).
Add check in old-configure as well. Reformat the check to use a maximum of 80
columns in the wscript.
Signed-off-by: Stefano Pigozzi <stefano.pigozzi@gmail.com>
This code is shared between input.conf parser and option parser. Until
now, the performance didn't really matter. But I want to use this code
for JSON parsing too, and since JSON will have to be parsed a lot, it
should probably try to avoid realloc'ing too much.
This commit moves parsing of C-style escaped strings into a common
function, and allows using it in a way realloc can be completely
avoided, if the already allocated buffer is large enough.
Apparently this can be really useful when being paranoid and trying to
avoid too much malloc/realloc, since it can be used to appending into a
buffer (with transparent realloc only if the buffer is too small).
Set the flag CODEC_FLAG_OUTPUT_CORRUPT by default. Note that there is
also CODEC_FLAG2_SHOW_ALL, which is older, but this seems to be ffmpeg
only.
Note that whether you want this enabled depends on the user. Some might
prefer that only good frames are output, while others want the decoder
to try as hard as possible to output _anything_. Since mplayer/mpv is
rather the kind of player that tries hard instead of being "clever", set
the new default to override libavcodec's default.
A nice way to test this is switching video tracks. Since mpv doesn't
wait for the next key frame, it'll start feeding the decoder with a
packet from the middle of the stream.
The Haali Matroska splitter is basically the reference implementation
for this crap, and it knows only:
application/vnd.ms-opentype
application/x-font-ttf
application/x-truetype-font
Two of them were missing in our code. One of them, "application/x-font",
is probably plain incorrect, but I can't really tell.
Also see: http://www.cccp-project.net/beta/test_files/fontsample.mkv
This is probably useful.
Note that this includes a small, stupid hack to prevent loading of the
config file if vf_lavfi is not available. The profile by default uses
vf_lavfi, and the config parser will output errors if vf_lavfi is not
available.
As another caveat, we install the example profile even if encoding is
disabled (though we don't load it, since this would print errors).
stream_read_complete() fails if the file is larger than the requested
maximum size. But input.c didn't check for this case, and no indication
that something went wrong was printed.
This reverts commit 877303aaa9.
The OpenGL 2.1 fallback for vo_opengl didn't work. Two things come
together: 1. trying to create an OpenGL 3.0 context will fail with
a GLXBadFBConfig error, and 2. X errors are fatal by default. Since
the reverted commit removed the X error handler, the mpv process was
killed, instead of continuing for the fallback.
(Note that this commit is not an exact inverse commit, since mp_msg
changed, but it does about the same thing.)
Apparently, Matroska packs TrueHD packets in a way lavc doesn't expect.
This broke decoding of some files [1] completely. A short look at the
libavcodec parser shows that parsing this ourselves would probably be
too much work, so make use of the libavcodec parser API.
[1] http://www.cccp-project.net/beta/test_files/mzero_truehd_sample.mkv
query_format was setting state even if wasn't the correct thing to do. Somehow
it worked by pure luck (until commit e6e6b88b6d).
Fix the initialization by setting state inside of reconfig.
Currently KDE will copy a media file into a temporary folder instead of
trying to stream it if a KIO slave location file is started. This change
will tell KDE to make mpv try to directly play the file. Perhaps the
proper flags should be added according to the individual enabled
features of the build but I suggest that be for the future.
Signed-off-by: wm4 <wm4@nowhere>
Hide --enable variants from [autodetect]'ed options and --enable/--disable
variants for [enable]'d/[disable]'d options. The hidden options are still
usable, just hidden for more readability.
If the utf8 string used to create the NSString for title was invalid utf8,
-stringWithUTF8String returned nil and triggered an assertion in Cocoa's
framework code.
Sanitize the utf8 string and if the sanitation wasn't enough just avoid
crashing by not setting a title.
Fixes#406
This applies the usual logic of resetting stream selections to default
when switching to a file with a different track layout. (This is to
prevent selecting random streams.)
Also, make sure that a track can't be selected twice. While this might
work in some situations, it certainly won't work with subtitles demuxed
from a stream.
Fixes#425.
This is relatively hacky, but it's Christmas, so it's ok. This does two
things: 1. allow selecting two subtitle tracks, and 2. include a hack
that renders the second subtitle always as toptitle. See manpage
additions how to use this.
Normally, there can be only one demuxer stream active for each demuxer
of an external file, but this assumption will be broken for multiple
subtitles support.
For some reason, this checked whether there are external tracks at all
before doing any seeks. Possibly this was to avoid multiple
get_main_demux_pts() calls, but calling this multiple times shouldn't be
too bad.
Normally we shouldn't load these files. But for some reason it was added
in commit b784346e some years ago, and disabling this hack would
probably be an inconvenience. So just print a warning.
The only thing that used mp_load_per_file_config() was inside
configfiles.c too, so remove the declaration from core.h and move the
function before its use.
Code for loading "[vo.vdpau]" profiles and similar. The messages printed
on loading change, but other than that, everything should behave about
the same.
Including the "." in the returned extension was too inconvenient. I
think originally, the semantics were supposed to work like in Python,
but screw this.
Also, return NULL instead of "" on failure (which is what its only user
actually seems to expect).
How embarrassing...
This code is inactive for all VOs other than vo_vdpau. For vo_vdpau,
this caused various issues, such as stuttering after about an hour of
running mpv; see github issue #403.
Note that this will print a difference even with perfect sync, because
the code queues the frames _between_ vsync, probably for error margin
(though I don't understand why it uses the exact values chosen).
I find this annoying. It's the reason common/version.c exists at all.
options.c did this for the user agent, which contains the version
number. Because not including version.h means you can't build the user
agent and use it in mp_default_opts anymore, do something rather awkward
in main.c to initialize the default user agent.
Make it so --msglevel extends previous --msglevel uses, instead of
overwriting them. Do this by literally appending the --msglevel option
value to the previous one.
There's a single mp_msg() in path.c, but all path lookup functions seem
to depend on it, so we get a rat-tail of stuff we have to change. This
is probably a good thing though, because we can have the path lookup
functions also access options, so we could allow overriding the default
config path, or ignore the MPV_HOME environment variable, and such
things.
Also take the chance to consistently add talloc_ctx parameters to the
path lookup functions.
Also, this change causes a big mess on configfiles.c. It's the same
issue: everything suddenly needs a (different) context argument. Make it
less wild by providing a mp_load_auto_profiles() function, which
isolates most of it to configfiles.c.
The TV code pretends to be part of stream/, but it's actually demuxer
code too. The audio_in code is shared between the TV code and
stream_radio.c, so stream_radio.c needs a small hack until stream.c is
converted.
This is pretty nasty, because FFmpeg/Libav is yet another library with a
global message callback. We do something with mutexes trying to get it
done, but of course we can't actually solve this problem. If more than
one library in a process use FFmpeg/Libav, only one of them will get log
messages.
Always pass around mp_log contexts in the option parser code. This of
course affects all users of this API as well.
In stream.c, pass a mp_null_log, because we can't do it properly yet.
This will be fixed later.
Until now, there were two functions to add input sources (stuff like
stdin input, slave mode, lirc, joystick). Unify them to a single
function (mp_input_add_fd()), and make sure the associated callbacks
always have a context parameter.
Change the lirc and joystick code such that they take store their state
in a context struct (probably worthless), and use the new mp_msg
replacements (the point of this refactoring).
Additionally, get rid of the ugly USE_FD0_CMD_SELECT etc. ifdeffery in
the terminal handling code.
The mp_select_decoders() function doesn't have a log context (and I
don't want to give it one), so get rid of the mp_msg error message by
enhancing the semantics such that the syntax error is replaced by a new
feature. Now doing "--ad=something" will enable all decoders in the
"something" module, same as "--ad=something:*". Pretty useless, but gets
rid of the annoyance.
Remove the nonsensical print_lock too.
Things that are called from the option validator are not converted yet,
because the option parser doesn't provide a log context yet.
This ended up a little bit messy. In order to get a mp_log everywhere,
mostly make use of the fact that va_surface already references global
state anyway.
This removes the messages printed on unknown pixel format messages.
Passing a mp_log to them would be too messy. Actually, this is a good
change, because in the past we often had trouble with these messages
printed too often (causing terminal spam etc.), and printing warnings or
error messages on the caller sides is much cleaner.
vd_lavc.c had a change earlier to print an error message if a decoder
outputs an unsupported pixel format.
This requires the caller to provide a mp_log in order to see error
messages. Unfortunately we don't do this in most places, but I guess we
have to live with it.
lcms2 has a global message callback for error reporting. If you don't
set this, these error messages are silently thrown away. I think we
still want the error messages, so we have to do dumb stuff to avoid
clashes. This doesn't handle the case if another library in the same
process sets the message callback, but at least this should exclude
possible memory errors when running multiple instances of mpv.
This has similar problems as the ALSA message callback, though in theory
we could use the Display handle to find the right mpv instance from the
global callback. It still wouldn't work if another library happens to
set the error handler at the same time. There doesn't seem much of an
advantage overriding the error handler (though it used to be required),
so remove it.
This could output additional, potentially useful error messages. But the
callback is global and not library-safe, and would require us to add
additional state. Remove it, because it's obviously too much of a pain.
Also, it seems ALSA prints stuff to stderr anyway.
I don't know under which circumstances this can error (other than a
broken format string). It seems it won't return an error code on I/O
errors, so maybe broken format strings are the only case. Either way,
don't continue if an error is returned.
We have certain race conditions coming from doing multiple fprintf()
calls (setting up colors etc.). I'm not sure whether it would be worth
changing to code such that we do only one fprintf() call (and assume
this synchronizes access), but considering it would be hard to do
(Windows compatibility, ...), and that stdio uses per FILE locks anyway,
this is simpler and probably not less efficient. Also, there's no
problem handling the weird statusline special case this way.
Note that mp_msg_* calls which are silent won't acquire the lock, and
acquiring the lock happens on actual output only (which is slow and
serialized anyway).
Basically, reimplement --msglevel. Instead of making the new msg code
use the legacy code, make the legacy code use the reimplemented
functionality.
The handling of the deprecated --identify switch changes. It temporarily
stops working; this will be fixed in later commits.
The actual sub-options syntax (like --msglevel-vo=...) goes away, but I
bet nobody knew about this or used this anyway.
Instead of making msg.c an ifdef hell for unix vs. windows code, move
the code to separate functions defined in terminal-unix.c/terminal-
win.c.
Drop the code that selects random colors for --msgmodule prefixes.
So you can pass a command as list of strings (each item is an argument),
instead of having to worry about escaping and such.
These functions also take an argument for the default command flags. In
particular, this allows setting saner defaults for commands sent by
program code.
Expose this to Lua as mp.send_commandv command (suggestions for a better
name welcome). The Lua version doesn't allow setting the default command
flags, but it can still use command prefixes. The default flags are
different from input.conf, and disable OSD and property expansion.
This comes with a real change in behavior: now the signal handler is set
only when the terminal input code is active (e.g. not with
--no-consolecontrols), but this should be ok.
Remove these because I'm too lazy to convert them to proper
STREAM_CTRLs. Considering that probably nobody uses radio://, caring
about this is a complete waste of time. I will add these commands back
if someone asks for them, but I don't expect this to happen.
Apparently this has been broken for a year or so. The were three
reasons for the breakage here:
1. The window dragging hack prevented any DOWN event from
passing through since it always returned before we even got
the button.
2. The window style had CS_DBLCLKS in its flags, so we did not
get any DOWN events when the OS had detected a double click
(instead expecting us to handle a DBL event).
3. We never sent any mouse buttons when mouse movement handling
was disabled.
request_channels has been deprecated for years (request_channel_layout
is the replacement), but it appears it's still needed despite the
deprecation at least on older libavcodec versions.
So still set request_channels, but to it with the avoption API, which
hides the deprecation warning. This should also prevent mpv getting
trashed when libavcodec happens to bump its major version.
In my opinion, config.h inclusions should be kept to a minimum. MPlayer
code really liked including config.h everywhere, though, even in often
used header files. Try to reduce this.
Since m_option.h and options.h are extremely often included, a lot of
files have to be changed.
Moving path.c/h to options/ is a bit questionable, but since this is
mainly about access to config files (which are also handled in
options/), it's probably ok.
The tmsg stuff was for the internal gettext() based translation system,
which nobody ever attempted to use and thus was removed. mp_gtext() and
set_osd_tmsg() were also for this.
mp_dbg was once enabled in debug mode only, but since we have log level
for enabling debug messages, it seems utterly useless.
Nothing actually used the returned length. Since the remaining time can
easily become 0 or negative (e.g. incorrectly estimated file duration),
the time_remaining function still needs 2 return values, though.
"-" could skip optional arguments. I think this was a pretty bad idea,
because it introduced a weird special case.
I'll remove the special syntax, but keep compatibility for the "seek"
and "screenshot" commands.
If input is not waiting for select(), writing to the wakeup pipe is
wasteful, and, if there are many wakeups, might even block the wakeup
threads if the pipe gets full.
However, if it's waiting for select(), the wakup pipe must be used to
unblock the waiting thread. Actually there's a small race condition: we
might determine that the main thread is in select(), and write to the
wakeup pipe (whether we do this while unlocked or locked doesn't really
matter). Then, the main thread might leave select() before reading from
the wakup pipe. This should be harmless, because at worst more wakeups
than needed happen, but never fewer.
In order to use bare mp_log contexts, I find myself creating dummy
structs (with a single "log" field) to use the MP_ERR() etc. macros,
which hardcode the idiom that a context struct has a log field. On the
other hand, just using mp_msg_log() is too much typing (and I want to
rename it to mp_msg() when the transition is done), so it seems nice to
have message printing macros that use mp_log directly.
There's no reason why we should do this. For some reason, the existing
code reset terminal handling to default after unloading a file, just to
initialize it again when loading a new file.
Might be related to github issue #412, although I don't think it helps,
since the default SIGTERM handler _should_ kill the mpv process.
(It's still a nice simplification, though.)
The OSD style settings depend on the PlayRes, simply because all style
values are implicitly scaled by the PlayResY in libass. Also, the OSC
changes the PlayResY in certain situations, so something could go wrong.
But not sure if this actually matters in practice.
This simplifies things, although it is slightly less efficient (probably
uses a bit more memory).
This also happens to fix that the OSC dropped the libass cache on every
frame.
This doesn't have much value. It can't be accessed by anything else than
the actual subtitle renderer (sd_ass.c). sd_ass.c could create the
renderer itself, except that we apparently want to save memory (and some
font loading time) when using ordered chapters or multiple subtitle
tracks.
The end of the current segment will be the end of the file if there is
no next segment. Normally, this didn't matter much, since UNIX files
allow seeking past the end of the file. But when opening files from
HTTP, this would print confusing error messages. So explicitly check for
EOF before trying to read a segment.
Add the mp_get_user_path() function, and make it expand special path
prefixes. Use it for some things in mpv which take filenames
(--input-config, --screenshot-template, opengl icc-profile suboption).
This allows accessing files in the mpv config dir without hardcoding the
config path by prefixing the path with ~~/. Details see manpage
additions.
Before this, they were displayed forever. Since some dvd screens seem
not to allow escaping from the still frame using the menu, this could
get you stuck forever.
On dvdnav, caching kind of works but not really. (Not our fault, at
least not fully. It's due to libdvdnav being slightly misdesigned; see
previous commit for some explanations.)
The TV code is implemented in the demuxer, and the stream implementation
is just a wrapper, so caching makes no sense here.
No idea why this was disabled. It was in the original MPlayer code,
which doesn't make much sense to me, because using the MPlayer stream
cache seems 100% broken due to design issues.
Not sure how this should work. Using libdvdnav with a large FIFO doesn't
make sense either: data and control commands use the same stream, so if
you want to send input to libdvdnav, you just have to read from
libdvdnav all the time to get a reaction, which is not compatible with
maintaining a buffer that could remain full for a long time.
I have no idea either whether this improves or worsens anything, though
it might be more correct.
EOF is a special case. Normally, the reader will block until the cache
thread has new data. Obviously we don't want to do this on EOF, because
we'd potentially block forever. On the other hand, EOF will put the
cache thread into a waiting state, so if EOF recovers, this will happen
at a "later" point. This is bad if there is some kind of external event
that ends the EOF condition. In this case, a steram_read() call would
still return EOF. Make it so that the reader waits at least for one
iteration of the cache trying to rad a new block.
Also adjust some debug messages to not print file positions in hex.
There was already something similar in the code that did the actual
seek, but I think seeking to the same position could still trigger an
actual seek due to weid interaction with the buffer.
Use the scaled video size (i.e. as shown on the window) as reference for
zoom. This is the easiest way to fix different width/height scale
factors as they happen when zooming video with a pixel aspect ratio
other than 1:1.
Also fix the unscaled mode, so that it 1. doesn't scale even with
--video-zoom, and 2. doesn't scale by small amounts when the video is
cropped by making the window smaller than the video.
Since dvb input is not seekable anyway, this would probably just flush
the stream buffer, but at least it'll avoid a bogus message about
seeking errors.
This code tried to pass a still monotonic (even if not strictly
monotonic) PTS to the player, but as a result it remained stuck on
the PTS before a reset (since the PTS was lower).
This readds a more or less completely new dvdnav implementation, though
it's based on the code from before commit 41fbcee. Note that this is
rather basic, and might be broken or not quite usable in many cases.
Most importantly, navigation highlights are not correctly implemented.
This would require changes in the FFmpeg dvdsub decoder (to apply a
different internal CLUT), so supporting it is not really possible right
now. And in fact, I don't think I ever want to support it, because it's
a very small gain for a lot of work. Instead, mpv will display fake
highlights, which are an approximate bounding box around the real
highlights.
Some things like mouse input or switching audio/subtitles stream using
the dvdnav VM are not supported.
Might be quite fragile on transitions: if dvdnav initiates a transition,
and doesn't give us enough mpeg data to initialize video playback, the
player will just quit.
This is added only because some users seem to want it. I don't intend to
make mpv a good DVD player, so the very basic minimum will have to do.
How about you just convert your DVD to proper video files?
The code to remove a section from the active section array wasn't
correct (it should have tried to copy the elements in reverse), so just
replace it with a macro that does the intended thing.
Add some debug output to print the section stack.
Until now, the player didn't care to drain frames on video reconfig.
Instead, the VO was reconfigured (i.e. resized) before the queued frames
finished displaying. This can for example be observed by passing
multiple images with different size as mf:// filename. Then the window
would resize one frame before image with the new size is displayed. With
--vo=vdpau, the effect is worse, because this VO queues more than 1
frame internally.
Fix this by explicitly draining buffered frames before video reconfig.
Raise the display time of the last frame. Otherwise, the last frame
would be shown for a very short time only. This usually doesn't matter,
but helps when playing image files. This is a byproduct of frame
draining, because normally, video timing is based on the frames queued
to the VO, and we can't do that with frames of different size or format.
So we pretend that the frame before the change is the last frame in
order to time it. This code is incorrect though: it tries to use the
framerate, which often doesn't make sense. But it's good enough to test
this code with mf://.
This gets rid of the vf_vo pseudo-filter. It ends the idea of MPlayer's
architecture that the VO is just a (terminating) video filter. It didn't
really work for us with respect to video timing (the "end" of the video
chain isn't really made for video timing, and making it do so would be
awkward), and now we're removing it entirely. We will be able to fix
some things, such as properly draining video on reconfiguration.
I don't think this has any reason to exist. It's likely that this used
to be required by the old direct rendering infrastructure. (See
git blame output.)
This should help fixing some issues (like not draining video frames
correctly on reinit), as well as decoupling the decoder, filter chain,
and VO code.
I also wanted to make the hardware video decoding fallback work properly
if software-only video filters are inserted. This currently has the
issue that the fallback is too violent, and throws away a bunch of
demuxer packets needed to restart software decoding properly. But
keeping "backup" packets turned out as too hacky, so I'm not doing this,
at least not yet.
Libav 9 still uses the unprefixed PIX_FMT_... symbols, but they will
probably be removed some time in the future.
There are some other deprecations we have yet to take care of, but
there are no clear replacements yet.
The previous RING_BUFFER_COUNT value, 64, would have ao_wasapi buffer 64
frames of audio in the ring buffer; a delay of 1280ms, which is clearly
overkill for everything. A value of 8 buffers 8 frames for a total of
160ms.
When get_space was converted to returning samples instead of bytes, a
unit type mismatch in get_delay's calculation returned bogus values. Fix
by converting get_space's value back to bytes.
Fixes playback with ao_wasapi when reaching EOF, or seeking past it.
There are 3 users of the image format option type: demux_raw,
vf_format, vf_noformat. Allow the hwaccel formats (like vdpau etc.)
in general, so that the filters can use it. This won't work for
demux_raw, so explicitly reject these formats there.
Remove the inconsistent, duplicated, and insufficient scale filter
insertion code, and do it in one place instead. This also compensates
for the earlier removal of vf_match_csp() (which was in fact duplicated
code).
The algorithm to determine where to insert a filter etc. is probably the
same, though it also comes with some changes that should make debugging
easier when trying to figure out why a chain is failing to configure.
Add an "in" pseudo filter, which makes insertion of conversion filters
easier. Also change the vf->reconfig signature. At a later point, I'll
probably change format negotiation such that the generic filter code
will choose the output format, so having separate in and out params will
be useful.
Reason: I never liked it being recursive. Generally, this seems to
cause more problems than trouble, and is less flexible for access
outside of the chain.
I don't think we need these flags anymore. Simplify the code and get rid
of the vf_format struct.
There still is the vf_format.configured field, but this can be replaced
by checking for a valid image format.
This adds vf_chain, which unlike vf_instance refers to the filter chain
as a whole. This makes the filter API less awkward, and will allow
handling format negotiation better.
This function improves automatic filter insertion, but this really
should be done by the generic filter code. Remove vf_match_csp() and all
code using it as preparation for that.
This commit temporarily makes handling of filter insertion worse for
now, but it will be fixed with the following commits.
If sys/soundcard.h is actually linux/soundcard.h then it supports only OSSv3
API. This may happen when OSSLIBDIR == /usr while forgetting to replace
sys/soundcard.h from glibc.
However, after fa620ff waf prefers native implementation which is inferior
on Linux. To fix try making waf prefer oss-audio-4front. It's quite unusual
to have 4Front OSS installed where native implementation is superior, anyway.
Signed-off-by: bugmen0t <@>
Make the false positives path also undef the 4Front define.
Signed-off-by: Stefano Pigozzi <stefano.pigozzi@gmail.com>
Fixes#396
Bug introduced by commit 6fb020f5. It doesn't always happen, since it is
caused by the playloop and cocoa UI code running in separate threads.
Fixes#398.
Uncompressed rar archives can be transparently opened, but the filename
the player doesn't have the direct filename (but something starting
with rar://... instead). This will lead to external subtitles not
being loaded.
This doesn't handle multi-volume rar files, but in that cases just use
the --autosub-match=fuzzy option.
Fixes#397 on github.
If only coreaudio was activativated and not cocoa, the build failed for
missing CoreFoundation.
Signed-off-by: Stefano Pigozzi <stefano.pigozzi@gmail.com>
Fixes#395
They didn't do anything.
vf_screenshot.c actually did release the previous image, but that's not
really required. At worst you could take a screenshot and get an old
frame when there's no new frame yet.
The --flip option flipped the image upside-down, by trying to use VO
support, or if not available, by inserting a video filter. I'm not sure
why it existed. Maybe it was important in ancient times when VfW based
decoders output an image this way (but even then, flipping an image is a
free operation by negating the stride).
One nice thing about this is that it provided a possible path for
implementing video orientation, which is a feature we should probably
support eventually. The important part is that it would be for free for
VOs that support it, and would work even with hardware decoding.
But for now get rid of it. It's useless, trivial, stands in the way, and
supporting video orientation would require solving other problems first.
In particular, this disables mpeg4. There are some files out there that
use GMC, a usually rarely used and ineffective feature, which is not
supported by most hardware decoders. In these cases the hw decoder
outputs garbage, while software decoding works perfectly fine. We can't
really fallback to software decoding in these cases, because we don't
know that something is wrong in the first place. I can't see any
advantages of hw decoding of mpeg4, so it's better to disable it.
This can be reproduced with:
mpv short.wav -af 'lavfi="aecho=0.8:0.9:5000|6800:0.3|0.25"'
An audio file that is just 1-2 seconds long should play for 8-9 seconds,
which audible echo towards the end.
The code assumes that when playing with AF_FILTER_FLAG_EOF, the filter
will either produce output, or has all remaining data flushed. I'm not
really sure whether this really works if there are multiple filters with
EOF handling in the chain. To handle it correctly, af_lavfi should retry
filtering if 1. EOF flag is set, 2. there were input samples, and 3. no
output samples were produced. But currently it seems to work well enough
anyway.
The new signature is actually closer to how it actually works, and
someone who is not familiar to the API and how it works might make fewer
fatal mistakes with the new signature than the old one. Pretty weird.
Do this to sneak in a flags parameter, which will later be used to flush
remaining data of at least vf_lavfi.
This will make af_channels output a channel layout that is compatible
with any destination layout. Not sure if that's a good idea though,
since the way the AO choses a layout is perhaps less predictable. On the
other hand, using the old MPlayer standard layouts doesn't make much
sense either. We'll see whether this improves or breaks someone's use
case.
Apparently this stopped working after some planar changes (broken format
negotiation). Radically change option parsing in an incompatible way.
Suggest alternatives to this filter, since it barely has any importance
anymore.
Normally, audio decoder don't have a decoder delay, so the code was
fine. But FFmpeg supports multithreaded decoding for some audio codecs,
which introduces such a delay.
The delay means that we won't get decoded audio for the first few
packets, and that we need to do something to get the trailing audio
still buffered in the decoder when reaching EOF.
Two changes are needed to deal with the delay:
- If EOF is reached, pass a "flush" packet to the decoder to return the
buffered audio. Such a flush packet is automatically setup when
calling mp_set_av_packet() with a NULL packet.
- Use the PTS returned by the decoder, instead of the packet's. This is
important to get correct timestamps for decoded audio. Ignoring this
would result into offsetting the audio playback time by the decoder
delay. Note that we can still use the timestamp of the first packet
to get the timestamp for the start of the audio.
If the timebase is set, it's used for converting the packet timestamps.
Otherwise, the previous method of reinterpret-casting the mpv style
double timestamps to libavcodec style int64_t timestamps is used.
Also replace the kind of awkward mp_get_av_frame_pkt_ts() function by
mp_pts_from_av(), which simply converts timestamps in a way the old
function did. (Plus it takes a timebase parameter, similar to the
addition to mp_set_av_packet().)
Note that this should not change anything yet. The code in ad_lavc.c and
vd_lavc.c passes NULL for the timebase parameters. We could set
AVCodecContext.pkt_timebase and use that if we want to give libavcodec
"proper" timestamps.
This could be important for ad_lavc.c: some codecs (opus, probably mp3
and aac too) have weird requirements about doing decoding preroll on the
container level, and thus require adjusting the audio start timestamps
in some cases. libavcodec doesn't tell us how much was skipped, so we
either get shifted timestamps (by the length of the skipped data), or we
give it proper timestamps. (Note: libavcodec interprets or changes
timestamps only if pkt_timebase is set, which by default it is not.)
This would require selecting a timebase though, so I feel uncomfortable
with the idea. At least this change paves the way, and will allow some
testing.
Don't bother explaining the sample format naming schema. The "ne" bit is
outdated anyway, and anyone who has to use this option will be able to
understand the naming schema just by looking at the names too.
This prevents waf from running test programs after compilation. A better
approach would be to only remove this option if the check actually errors,
but we are using this only for Lua anyway.
The vf_eq context contains a very large lookup table, and the method of
setting default values caused the vf_eq context to be included in the
compiled code.
vf_stereo3d now uses vf_lavfi, if mpv was compiled with libavfilter.
vf_swapuv is hereby undeprecated. It's too trivial to wrap it with
libavfilter, and it's also too useless that even typing this commit
message is not really worth the time to spend on it.
Just in case someone expects these are unchanged just because they're
not mentioned in changes.rst anywhere. Documenting all of these changes
would be too much work and not helpful either.
All filters now either use the generic option parser, or don't have
options. This finally finishes a transition started in 2003 (see git
commit 33b62af947).
Why are MPlayer devs so monumentally lazy? Sorry, but this takes the
cake. You had 10 years.
Mostly backwards compatible, we don't change much because we just want
to get rid of the legacy option string handling.
You can't pass an aspect as first argument anymore.
Apparently you can get this with: stereo3d=ab[2]{l,r}:sbs[2]{l,r}
So it seems the filter is redundant and can be removed.
Also see FFmpeg commit 2f11aa141a01.
Unfortunately, this forces filtering both luma and chroma, because
otherwise we'd have to deal with libavfilter's vf_noise weird handling
of YUV vs. RGB formats. Would we e.g. filter luma only, it would filter
red in RGB mode only, because it goes by component and there's no way to
distinguish YUV and RGB by just using the filter's options.
Also remove the ability to disable deinterlacing at runtime. You can
still disable deinterlacing at runtime by using the ``D`` key and its
automatical filter insertion/removal.
This will allow old filter to run libavfilter instead by calling
vf_lw_set_graph(), which turns the filter into a wrapper, using a given
libavfilter graph.
Later commits use that to automatically "reroute" a bunch of filters to
libavfilter. We want to get rid of the old MPlayer filter code, because
it's bad an unmaintained, but we still don't want to force everyone to
use vf_lavfi, so this solution will do for a while.
Previous API worked under the assumption that download_image is always called
after map_image. In practice this is true, but it's better to have a much
generic API that doesn't depend on the order in which the functions are called.
The hwdec driver can be loaded, even if it's not used (e.g. when playing
a file with no hardware decoding after one with it enabled).
Also, check whether dlimage is NULL. Since this will do call into the
native hwdec API, there's a chance a driver could fail doing this, it's
better to check the return value, even if this case currently can't
happen.
mpv was hardcoded to always consider the right Alt key as Alt Gr, but there
are parituclar combinations of platforms and keyboard layouts where it's more
convenient to treat the right Alt as a keyboard modifier just like the left
one.
Fixes#388
The harder work was done in the previous commits. After that this feature comes
out almost for free.
The only problem is I can't get the textures created with CGLTexImageIOSurface2D
to download properly, thus the code performs download using some CoreVideo APIs.
If someone knows why download of textures created with CGLTexImageIOSurface2D
doesn't work please contact me :)
This adds support for packed YUV formats (YUVY and UYVY) using the extension
GL_APPLE_rgb_422. While supporting this formats on their own is not that
important (considering most video is planar YUV) they are used for
interoperability with IOSurfaces.
Next commit will use this formats to render VDA hardware decoded frames through
IOSurface and OpenGL interoperability.
roaraudio has some sort of sndio emulation, but apparently its header
file is either blatantly broken, or an old version from the past. The
sio_onvol() function has the wrong return type (void instead of int),
and the SIO_DEVANY symbol is missing entirely. This broke the build,
because the configure check was successful anyway.
This allows vo_opengl to use GL_TEXTURE_RECTANGLE textures, either by
enabling it with the 'rectangle-textures' sub-option, or by having a
hwdec backend force it. By default it's off.
The _only_ reason we're adding this is because VDA can export rectangle
textures only.
We got a crash in libavutil when encoding with Y8 (GRAY8). The reason
was that libavutil was copying an Y8 image allocated by us, and expected
a palette. This is because GRAY8 is a PSEUDOPAL format. It's not clear
what PSEUDOPAL means, and it makes literally no sense at all. However,
it does expect a palette allocated for some formats that are not
paletted, and libavutil crashed when trying to access the non-existent
palette.
This is for key bindings that use multiple mouse buttons at once. (Yes,
this is weird, but MPlayer always had this feature, and apparently
there are people using it!)
Before this commit, clicking another mouse button while still holding
the previous mouse button forced the command bound to the previous
mouse button to be emitted. This is usually needed to make sure the
input consumer (the player and the OSC) stays in sync with the actual
mouse button state. If there's no command sent, the OSC in particular
would think the button is still held down. However, sending the command
is undesired behavior if you want to use these multiple-key binds.
Solve this by emitting commands in this situation only if a key down
command was sent earlier. Since mouse button key bindings are normally
executed on key-up only, this happens with special commands like
script_dispatch only (used by the OSD to track mouse buttons, but
also used for other OSC bindings).
See github issue #390.
This is not strictly needed anymore. (On the other hand, it's not really
possible to do hw decoding with vo_null, because the VO is still
responsible for opening the hw decoder API, but that's another story.)
There are some use cases for this. For example, you can use it to set
defaults of automatically inserted filters (like af_lavrresample). It's
also useful if you have a non-trivial VO configuration, and want to use
--vo to quickly change between the drivers without repeating the whole
configuration in the --vo argument.
The "run" input command does fork+exec to spawn child processes. But it
doesn't cleanup the child processes, so they are left as zombies until
mpv terminates. Leaving zombie processes around is not very nice, so
employ a simple trick to let pid 1 take care of this: we fork twice, and
when the first fork exits, the second fork becomes orphaned and becomes
pid 1's child. It becomes pid 1's responsibility to cleanup the process.
The advantage is that we don't need extra logic to cleanup the spawned
process, which could have an arbitrary lifetime.
This is e.g. described here: http://yarchive.net/comp/zombie_process.html
Also use _exit() instead of exit(). It's not really sane to run cleanup
handlers (atexit() etc.) inside a forked process.
This is needed so that new processes (created with fork+exec) don't
inherit open files, which can be important for a number of reasons.
Since O_CLOEXEC is relatively new (POSIX.1-2008, before that Linux
specific), we #define it to 0 in io.h to prevent compilation errors on
older/crappy systems. At least this is the plan.
input.c creates a pipe. For that, add a mp_set_cloexec() function (which
is based on Weston's code in vo_wayland.c, but more correct). We could
use pipe2() instead, but that is Linux specific. Technically, we have a
race condition, but it won't matter.
See the changes in input.rst for explanations.
Technically speaking, this also gets rid of some undefined behavior:
passing NULL as a vararg (execl()) is always a bug.
So e.g.
show_text abc#def
will now print "abc#def" instead of "#def". It's simpler, more
consistent with how ";" and other things are handled, and also
possibly avoids bothering the user with extra escaping.
The OSS checks were a big mess and quite buggy. This reimplementes them using
a declarative approach and clearly distinguishing between the various OSS
implementations. The code should now almost be auto-documenting.
We currently support the following implementations of OSS:
* platform-specific (with `sys/soundcard.h`)
* SunAudio (default on NetBSD and useable on OpenBSD even if we have sndio
support there).
* 4Front (default on FreeBSD)
Since now each OSS check also checks for the appropriate soundcard header,
remove the old soundcard check.
Many thanks to @bugmen0t for in depth info about all the BSDs.
Check #380 and #359 for more info on this commit.
PIX_FMT_* -> AV_PIX_FMT_* (except some pixdesc constants)
enum PixelFormat -> enum AVPixelFormat
Losen some version checks in certain newer pixel formats.
av_pix_fmt_descriptors -> av_pix_fmt_desc_get
This removes support for FFmpeg 1.0.x, which is even older than
Libav 9.x. Support for it probably was already broken, and its
libswresample was rejected by our build system anyway because it's
broken.
Mostly untested; it does compile with Libav 9.9.
This partially reverts commit 7d152965. It turns out that at least some
ALSA drivers (at least snd-hda-intel) report incorrect audio delay with
non-native sample rates, even if the sample rate is only very slightly
different from the native one.
For example, 48000Hz is fine on my hda-intel system, while both 8000Hz
and 47999Hz lead to a delay off by 40ms (according to mpv's A/V
difference display), which suggests that something in ALSA is
calculating the delay using the wrong sample rate.
As an additional problem, with ALSA resampling enabled, using
48001Hz/float/2ch fails, while 49000Hz/float/2ch or 48001Hz/s16/2ch
work. With resampling disabled, all these cases work obviously, because
our own resampler doesn't just refuse any of these formats.
Since some people want to use the ALSA resampler (because it's highly
configurable, supports multiple backends, etc.), we still allow enabling
ALSA resampling with an ao_alsa suboption.
These were confined to the video path, but resetting them even if no
video is available shouldn't really matter. Always resetting them makes
the logic easier to follow, I think.
Sometimes, vf_pullup hanged on seek. This was because it never was
properly reset. Old timestamps messed up the timestamp calculations,
which made the player show frames for a ridiculously long time, which is
perceived as pausing or hanging.
Recently, the check was moved, so it was printed only for source video
PTS (since that's easier, and filters should normally behave sane). But
it turns out it's trivial to print a warning in the filter case too by
reusing the code that normally checks for PTS forward jumps without
needing any additional code, so, fine, restore warning in this case.
The old ffmpeg vdpau support code uses separate vdpau pixel formats for
each decoder (pretty much because mplayer's architecture sucked), which
just gets into the way. Force the old decoder's output to IMGFMT_VDPAU,
and remove IMGFMT_IS_VDPAU() where we can remove it.
This should completely remove the differences betwene the old and new
vdpau decoder outside of the decoder.
PIX_FMT_VDA_VLD and PIX_FMT_VAAPI_VLD were never used anywhere. I'm not
sure why they were even added, and they sound like they are just for
compatibility with XvMC-style decoding, which sucks anyway.
Now that there's only a single vaapi format, remove the
IMGFMT_IS_VAAPI() macro. Also get rid of IMGFMT_IS_VDA(), which was
unused.
pthreads should be available anywhere. Even if not, for environment
without threads a pthread wrapper could be provided that can't actually
start threads, thus disabling features that require threads.
Make pthreads mandatory in order to simplify build dependencies and to
reduce ifdeffery. (Admittedly, there wasn't much complexity, but maybe
we will use pthreads more in the future, and then it'd become a real
bother.)
Introduce a mp_cmd_def struct to define commands, instead of using
mp_cmd for this. This way each command parameter can be a m_option,
instead of m_option plus some more stuff.
Define the ARG_ macros directly in terms of the OPT_ macros. Not sure if
this makes it easier to read (maybe not, even if it looks simpler), but
at least it makes it easier to add other option types.
Another idea was adding a name for each parameter (so you could have
named parameters), but not today.
The hr-seek code assumes that when seeking the demuxer, the first image
decoded after the seek will have a PTS exactly equal to the demuxer seek
target time, or before that target time. Incorrect timestamps,
implicitly dropped initial frames, or broken files/demuxers can all
break this assumption, and lead to hr-seek missing the seek target.
Generally, this is not much a problem (the user won't notice being off
by one frame), but it really shows when using the backstep feature. In
this case, backstepping would simply hang.
Add a simple hack that basically forces a minimal value for the --hr-
seek-demuxer-offset option (which is 0 by default) when doing a
backstep-seek. The chosen minimum value is arbitrary. There's no perfect
value, though in general it should perhaps be slightly longer than the
frametime, which the chosen value is more than enough for typical
framerates.
So, FFmpeg/Libav requires us to figure out video timestamps ourselves
(see last 10 commits or so), but the methods it provides for this aren't
even sufficient. In particular, everything that uses AVI-style DTS (avi,
vfw-muxed mkv, possibly mpeg4-in-ogm) with a codec that has an internal
frame delay is broken. In this case, libavcodec will shift the packet-
to-image correspondence by the codec delay, meaning that with a delay=1,
the first AVFrame.pkt_dts is not 0, but that of the second packet. All
timestamps will appear shifted. The start time (e.g. the time displayed
when doing "mpv file.avi --pause") will not be exactly 0.
(According to Libav developers, this is how it's supposed to work; just
that the first DTS values are normally negative with formats that use
DTS "properly". Who cares if it doesn't work at all with very common
video formats? There's no indication that they'll fix this soon,
either. An elegant workaround is missing too.)
Add a hack to re-enable the old PTS code for AVI and vfw-muxed MKV.
Since these timestamps are not reorderd, we wouldn't need to sort them,
but it's less code this way (and possibly more robust, should a demuxer
unexpectedly output PTS).
The original intention of all the timestamp changes recently was
actually to get rid of demuxer-specific hacks and the old timestamp
sorting code, but it looks like this didn't work out. Yet another case
where trying to replace native MPlayer functionality with FFmpeg/Libav
led to disadvantages and bugs. (Note that the old PTS sorting code
doesn't and can't handle frame dropping correctly, though.)
Bug reports:
https://trac.ffmpeg.org/ticket/3178https://bugzilla.libav.org/show_bug.cgi?id=600
Using --start with files that use DTS only, or which simply have broken
PTS timestamps, would incorrectly drop frames and possibly not execute
the seek correctly.
Add yet another heuristic to detect this. The intent is that --start and
hr-seeks in general should work correctly, but in order to keep things
fast, we still want to allow frame dropping during hr-seek if there are
no problems doing so. Do this by disabling frame dropping by default,
but re-enabling it if there are no problems found for a while. As a
consequence, --start might be somewhat slower, but normal user
interaction should remain as fast as before.
Note that there's something subtle about the added code: the
has_broken_packet_pts field is checked even before the first packet is
fed to dec_video.c, so the field must not be set to 0 right on start.
It's not initially set to 0 anyway, because the heuristic requires
decoding some images before enabling frame drop anyway.
Note 2: it's not clear whether frame dropping during hr-seek really
helps; I didn't benchmark it.
Previous code was using the values of the AudioChannelLabel enum directly to
create the channel bitmap. While this was quite smart it was pretty unreadable
and fragile (what if Apple changes the values of those enums?).
Change it to use a 'dumb' conversion table.
The d_video->pts field was a bit strange. The code overwrote it multiple
times (on decoding, on filtering, then once again...), and it wasn't
really clear what purpose this field had exactly. Replace it with the
mpctx->video_next_pts field, which is relatively unambiguous.
Move the decreasing PTS check to dec_video.c. This means it acts on
decoder output, not on filter output. (Just like in the previous commit,
assume the filter chain is sane.) Drop the jitter vs. reset semantics;
the dec_video.c determined PTS never goes backwards, and demuxer
timestamps don't "jitter".
Also get rid of the PTS check _after_ filters. This means if there's a
video filter which unsets PTS, no warning will be printed. But we assume
that all filters are well-behaved enough by now.
Refactor the PTS handling code to make it cleaner, and to separate the
bits that use PTS sorting.
Add a heuristic to fall back to DTS if the PTS us non-monotonic. This
code is based on what FFmpeg/Libav use for ffplay/avplay and also
best_effort_timestamp (which is only in FFmpeg). Basically, this 1. just
uses the DTS if PTS is unset, and 2. ignores PTS entirely if PTS is non-
monotonic, but DTS is sorted.
The code is pretty much the same as in Libav [1]. I'm not sure if all of
it is really needed, or if it does more than what the paragraph above
mentions. But maybe it's fine to cargo-cult this.
This heuristic fixes playback of mpeg4 in ogm, which returns packets
with PTS==DTS, even though the PTS timestamps should follow codec
reordering. This is probably a libavformat demuxer bug, but good luck
trying to fix it.
The way vd_lavc.c returns the frame PTS and DTS to dec_video.c is a bit
inelegant, but maybe better than trying to mess the PTS back into the
decoder callback again.
[1] https://git.libav.org/?p=libav.git;a=blob;f=cmdutils.c;h=3f1c667075724c5cde69d840ed5ed7d992898334;hb=fa515c2088e1d082d45741bbd5c05e13b0500804#l1431
These used the suffix _resync_stream, which is a bit misleading. Nothing
gets "resynchronized", they really just reset state.
(Some audio decoders actually used to "resync" by reading packets for
resuming playback, but that's not the case anymore.)
Also move the function in dec_video.c to the top of the file.
The code stopped at kAudioChannelLabel_TopBackRight and missed mapping for
5 more channel labels. These are in a completely different order that the mpv
ones so they must be mapped manually.
This includes the case when lavc decodes audio with more than 8
channels, which our audio chain currently does not support.
the changes in ad_lavc.c are just simplifications. The code tried to
avoid overriding global parameters if it found something invalid, but
that is not needed anymore.
When using PIC on x86 (eg with hardened toolchains) the ebx register is
reserverd and cannot be used in assembly code.
For vf_eq we allow the compiler to use memory as input.
For vf_noise we temoporarily borrow the ebp register.
This fixes#361.
Signed-off-by: Natanael Copa <ncopa@alpinelinux.org>
This was broken by the recent commits. Apparently realvideo timestamps
are severely mangled, and Matroska _of course_ doesn't have the sane,
umangled timestamps, but something unusable. The existing unmangling
code in demux_mkv.c didn't output proper timestamps either. Instead,
it was something weird that triggered sorting. Without sorting (it was
disabled by default recently), you'd get decreasing PTS warnings
In order to fix this, steal some code from libavcodec. Basically copy
the contents of rv34_parser.c (with some changes), which makes
everything magically work. (Maybe it would be better to use the
libavcodec parser API, but I don't want to do that just for this. An
alternative idea would be refusing to read files that have realvideo
tracks, and delegate this to demux_lavf.c, but maybe that's too redical
too.)
I wish I hadn't notice this...
Resampling with non-ancient ALSA setups works fine, so there is no
need to keep this around. Furthermore, as of writing, the default
builtin resampler used by many ALSA setups (taken from libspeex)
actually has higher quality than the default resampling modes of
avresample and swresample.
Apparently just 5 packets is not enough for the initial audio decode
(which is needed to find the format). The old code (before the recent
refactor) appeared to use 5 packets, but there were apparently other
code paths which in the end amounted to more than 5 packets being read.
The sample that failed (see github issue #368) needed 9 packets.
Fixes#368.
These packets have to be explicitly dropped, because usually libavcodec
uses 0-sized packets to flush delayed frames, meaning just passing
through these packets would have bad consequences.
Normally, libavformat doesn't output 0-sized packets anyway. But I don't
want to take any chances, so don't delete it, and just move it out of
the way to demux.c.
Incidentally this seems wrong in the configure as well because only gl-win32
depends on it instead of also making direct3d depend on this (as I did here).
If the value for --cache-on-pause is larger than --cache-min, and the
cache runs below --cache-on-pause, but above --cache-min, the logic
would demand to pause the player and then unpause immediately again.
This doesn't make much sense, and alternating the pause state in each
playloop iteration has negative consequences. Add an explicit check to
avoid this situation.
This means the code that tries to figure out the timestamp from
demuxer and decoder output is now all in dec_video.c. We set the
final timestamp on the returned image (mp_image.pts), as well as
the d_video->pts field.
The way the player uses d_video->pts field is still a bit messy. Maybe
this could be cleaned up later.
It appears PTS sorting was useful only for avi files (and VfW-muxed
mkv). Maybe it was historically also important for decoders with broken
or non-existent PTS reordering (win32 codecs?). But now that we handle
demuxers which outputs DTS only correctly, it just seems dead weight.
Disable it by default. The --pts-association-mode option is now forced
to always use the decoder's PTS value. You can still enable the old
default (auto) or force sorting. But we will probably remove this option
entirely at some point.
Make demux_mkv export timestamps at DTS when it's in VfW mode. This is
needed to get correct timestamps with the new default mode. demux_lavf
already does that.
This was needed to determine PTS from DTS, but the previous commits
make it unnecessary.
The builtin genpts hack was used for DVD, because libavformat's genpts
essentially went amok on DVD timestamp resets. See commit 65d87091 for
details.
Having the DTS directly can be useful for restoring PTS values.
The avi file format doesn't actually store PTS values, just DTS. An
older hack explicitly exported the DTS as PTS (ignoring the [I assume]
genpts generated non-sense PTS), which is not necessary anymore due to
this change.
Now the --no-correct-pts mode is like the normal mode, just with
different timestamp calculations. The semantics should be about the
same as before this commit.
Before this commit, this mode estimated the frame time by subtracting
successive packet PTS values. This is complete non-sense for video
codecs which use reordering. The code compensated frame times for these
non-sense using the FPS value, but confused the rest of the player with
non-sense jumping around timestamps. So, all in all this mode is not
very useful.
Repurpose this mode for fixed frame rate playback. This gives almost the
same behavior as the old mode with forced framerate (--fps option). The
result is simpler and often more robust.
Instead of passing the PTS as separate field, pass it as part of the
usual data structures. Basically, this removes strange artifacts from
the API. (It's not finished, though: the final decoded PTS goes through
strange paths, and filter_video() finally overwrites the decoded
mp_image's pts field with it.)
We also stop using libavcodec's reordered_opaque fields, and use
AVPacket.pts and AVFrame.pkt_pts. This is slightly unorthodox, because
these pts fields are not "really" opaque anymore, yet we treat them as
such. But the end result should be the same, and reordered_opaque is
marked as partially deprecated (it's not clear whether it's really
deprecated).
This normally shouldn't happen. It does happen with VfW-muxed mkv files,
and would normally happen with avi files (except that we force the
correct association mode using an explicit hack for avi).
This is usually prepended by warnings like:
Decreasing video pts: 0.125000 < 0.167000
and after the switch, there should be no warnings anymore.
Background: avi likes to use DTS for timestamps instead of PTS.
Basically, there are no proper timestamps in the file, only frame
numbers. And Matroska is insane and stores the made-up DTS instead
of a proper PTS. This will be handled properly later.
Original commit was implemented differently by @bugmen0t. The problem here was
that the waf API was called directly, instead of using our own check_cc (which
defaults, among other things, to non mandatory checks).
- without Utils.* always returns empty string
- subprocess doesn't need extra quoting for sh -c
- "source" is a bash'ism, not in POSIX sh
- most shell commands embed newline at the end
Apparently Cocoa precise scrolling generates a lot of spurious events with
a delta that is equal to 0.0. Make sure that they are discarded and not added
to the input queue.
Even though this only known to happen with Cocoa, I implemented this at core
level since it makes sense in general.
Fixes: #310
waf wrongly installs resources to BundleRoot/Resources instead of
BundleRoot/Contents/Resources. Will post a patch to them later. Either way,
this must worked around until the next waf patch release.
It seems like a good idea not to generate any additional network traffic
and wait times if we don't have to.
Also print the URL it's downloading from.
Note that if we require a newer waf release, there will be a problem.
Running ./bootstrap.py won't get the newest waf version anymore in case
the old version is in the source dir. Not sure how to handle this.
waf apparently only appends a pkgconfig flag if it doesn't already exist in
the lib storage. Since our configure often checks for multiple libraries in
one call we want to keep the flags as is. This is especially important to
always keep stuff like -lm in the right place.
This is not needed anymore, because we decided that the PAR of the
decoded video matters, and not the PAR of the filtered video that
arrives at the VO.
In Python sets are unordered, so iterating them after converting to a list
always leads to different results. The code iterated on them to collect all
the flags to pass to the compiler, and since the order of the flags changed,
waf would rebuild all of the C files. Seems like in Python 2 this worked as
expected by pure chance.
This commit stores the sets as lists, and converts them to sets when the set
operations are needed.
Fixes#363
When mpv is started with some video filters set (--vf is used), and
hardware decoding is requested, and hardware decoding would be possible,
but is prevented due to video filters that accept software formats only,
the fallback didn't work properly sometimes.
This fallback works rather violently: it tries to initialize the filter
chain, and if it fails it throws away the frame decoded using the
hardware, and retries with software. The case that didn't work was when
decoding the current packet didn't immediately lead to a new frame. Then
the filter chain wouldn't be reinitialized, and the playloop would stop
playback as soon as it encounters the error flag.
Fix this by resetting the filter error flag (back to "uninitialized"),
which is a rather violent, but somewhat working solution.
The fallback in general should perhaps be cleaned up later.
In theory, we can't really do this, because we don't know when a spdif
frame ends. Spdif transports compressed audio through audio setups that
were originally designed for PCM only (which includes the audio filter
chain, the AO API, most audio output APIs, etc.), and to reach this
goal, spdif pretends to be PCM. Compressed data frames are padded with
zeros, until a certain data rate is reached, which corresponds to a
pseudo-PCM format with 2 bytes per sample and 2 channels at 48000 Hz.
Of course an actual spdif frame is significantly larger than a frame
of the PCM format it pretends to be, so cutting audio data on frame
boundaries (as according to the pseudo-PCM format) merely yields an
incomplete and broken frame, not audio that plays for the desired
duration.
However, sending an incomplete frame might still be much better than the
current behavior, which simply ignores --end/--length (but still lets
the video end at the exact end time).
Should this result in trouble with real spdif receivers, this commit
probably has to be reverted.
If the --fps option was given (MPOpts->force_fps), the demuxer FPS value
was overwritten with the forced value. This was fine, since the demuxer
value wasn't needed anymore. But with the recent changes not to write to
the demuxer stream headers, we don't want to do this anymore. So
maintain the (forced/updated) FPS value in dec_video->fps.
The removed code in loadfile.c is probably redundant, and an artifact
from past refactorings.
Note that sub.c will now always use the demuxer FPS value, instead of
the user override value. I think this is fine, because it used the
demuxer's video size values too. (And it's rare that these values are
used at all.)
Now the actual decoder doesn't need to care about this anymore, and it's
handled in generic code instead. This simplifies vd_lavc.c, and in
particular we don't need to detect format changes in the old way
anymore.
The only reason why these structs were dynamically allocated was to
avoid recursive includes in stheader.h, which is (or was) a very central
file included by almost all other files. (If a struct is referenced via
a pointer type only, it can be forward referenced, and the definition of
the struct is not needed.) Now that they're out of stheader.h, this
difference doesn't matter anymore, and the code can be simplified.
Also sneak in some sanity checks.
This used to be needed to access the generic stream header from the
specific headers, which in turn was needed because the decoders had
access only to the specific headers. This is not the case anymore, so
this can finally be removed again.
Also move the "format" field from the specific headers to sh_stream.
Use sh_stream over sh_sub. Use dec_sub (and mpctx->d_sub) instead of the
stream header. This aligns the subtitle code with the recent audio and
video refactoring.
sh_sub still has the decoder context, though. This is because we want to
avoid reinit when switching segments with ordered chapters. (Reinit is
fast, except for creating the ASS_Renderer, which in turn triggers
fontconfig.) Not sure how much this matters, though, because the initial
segment switch will lazily initialize the decoder anyway.
This is similar to the sh_audio commit.
This is mostly cosmetic in nature, except that it also adds automatical
freeing of the decoder driver's state struct (which was in
sh_video->context, now in dec_video->priv).
Also remove all the stheader.h fields that are not needed anymore.
This drops the --pp option, which was probably broken for a while. The
option automatically inserted the "pp" filter. The value passed to it
was ignored (which is probably broken, it always selected maximal
quality).
Inserting this filter can be done simply with --vf=pp, so this is not
needed anymore.
This means most code accessing this struct must now include hwdec.h
instead of dec_video.h. I just put it into dec_video.h at first because
I thought a separate file would be a waste, but it's more proper to do
it this way, as there are too many files which include dec_video.h only
to get the mp_hwdec_info definition.
This was printed before something was decoded, and thus was not really
correct. Also, this code is hilariously broken:
/* Assume FOURCC if all bytes >= 0x20 (' ') */
if (sh_audio->format >= 0x20202020)
mp_msg(MSGT_IDENTIFY, MSGL_INFO,
"ID_AUDIO_FORMAT=%.4s\n", (char *)&sh_audio->format);
Time to kill it.
This information can be accessed through properties instead.
sh_audio is supposed to contain file headers, not whatever was decoded.
Fix this, and write the decoded format to separate fields in the decoder
context, the dec_audio.decoded field. (Note that this field is really
only needed to communicate the audio format from decoder driver to the
generic code, so no other code accesses it.)
Move all state that basically changes during decoding or is needed in
order to manage decoding itself into a new struct (dec_audio).
sh_audio (defined in stheader.h) is supposed to be the audio stream
header. This should reflect the file headers for the stream. Putting the
decoder context there is strange design, to say the least.
Apparently some packaging systems (homebrew does this for example) change
`sys.path` before they run any Python script to ensure that only the correct
Python modules can be loadable.
We need to make sure cwd is in `sys.path` since we need to load
`wscript_build.py` from there.
Simply override the cprogram Task's __str__ method with our own implementation.
This is way easier on the eyes when compiling mpv during development, since
warnings are not pushed outside of your average screenful of content.
vo_opengl always loads the hwdec backend lazily, so hwdec_request_api()
has to be called to possibly load it. This makes vf_vavpp work with
software decoding. (Hardware decoding loads the backend before the
filter is initialized, so this case is different.)
Also, the VFCTRL_GET_HWDEC_INFO call doesn't need to be checked. If it
fails, the info will be left blank.
This initialized only the load_api and load_api_ctx fields, and left the
other fields as they were. This failed with vf_vavpp, which assumed all
fields are initialized.
This commit adds a new build system based on waf. configure and Makefile
are deprecated effective immediately and someday in the future they will be
removed (they are still available by running ./old-configure).
You can find how the choice for waf came to be in `DOCS/waf-buildsystem.rst`.
TL;DR: we couldn't get the same level of abstraction and customization with
other build systems we tried (CMake and autotools).
For guidance on how to build the software now, take a look at README.md
and the cross compilation guide.
CREDITS:
This is a squash of ~250 commits. Some of them are not by me, so here is the
deserved attribution:
- @wm4 contributed some Windows fixes, renamed configure to old-configure
and contributed to the bootstrap script. Also, GNU/Linux testing.
- @lachs0r contributed some Windows fixes and the bootstrap script.
- @Nikoli contributed a lot of testing and discovered many bugs.
- @CrimsonVoid contributed changes to the bootstrap script.
This was forgotten when the parser for mplayer2 EDL files was removed.
Change the header of the mpv EDL format to include a '#', so a naive
parser could skip the header as comment. (Maybe this is questionable;
on the other hand, if it can be simpler, why not.)
Also, strip the header in demux_edl.c before passing on the data, so the
header check doesn't need to be duplicated in tl_mpv_edl.c.
Edit Decision Lists (EDL) allow combining parts from multiple source
files into one virtual file. MPlayer had an EDL format (which sucked),
which mplayer2 tried to improve with its own format (which sucked). As
logic demands, mpv introduces its very own format (which sucks).
The new format should actually be much simpler and easier to use, and
its implementation is simpler and smaller too.
The intention of the existing code was trying to match demuxer-reported
stream IDs, instead of using possibly arbitrary ordering of the frontend
track list. But EDL files can consist of quite different files, for
which trying to match the stream IDs doesn't always make sense.
This didn't have any consequences, other than suddenly reinitializing
video when it works again (such as with EDL timeline mixing video and
audio-only files).
The AF control commands used an elaborate and unnecessary organization
for the command constants. Get rid of all that and convert the
definitions to a simple enum. Also remove the control commands that
were not really needed, because they were not used outside of the
filters that implemented them.
And by "cleanup", I mean "remove". Actually, only remove the parts that
are redundant and doxygen noise. Move useful parts to the comment above
the function's implementation in the C source file.
Some decoders used to read packets and decode data when calling
resync_audio_stream(). This required a special case in mp_seek() for
audio. (A comment mentions liba52, which is long gone; but until
recently ad_mpg123.c actually exposed this behavior.)
No decoder does this anymore, and resync_audio_stream() works similar
as resync_video_stream(). Remove the special case.
The priv struct is now allocated by talloc in stream.c. It doesn't need
to be manually freed, and using free() instead of talloc_free() probably
crashes.
When the decoder detects a format change, it overwrites the values
stored in sh_audio (this affects the members sample_format, samplerate,
channels). In the case when the old audio data still needs to be
played/filtered, the audio format as identified by sh_audio and the
format used for the decoder buffer can mismatch. In particular, they
will mismatch in the very unlikely but possible case the audio chain is
reinitialized while old data is draining during a format change.
Or in other words, sh_audio might contain the new format, while the
audio chain is still configured to use the old format.
Currently, the audio code (player/audio.c and init_audio_filters) access
sh_audio to get the current format. This is in theory incorrect for the
reasons mentioned above. Use the decoder buffer's format instead, which
should be correct at any point.
Commit 22b3f522 not only redid major aspects of audio decoding, but also
attempted to fix audio format change handling. Before that commit, data
that was already decoded but not yet filtered was thrown away on a
format change. After that commit, data was supposed to finish playing
before rebuilding filters and so on.
It was still buggy, though: the decoder buffer was initialized to the
new format too early, triggering an assertion failure. Move the reinit
call below filtering to fix this.
ad_mpg123.c needs to be adjusted so that it doesn't decode new data
before the format change is actually executed.
Add some more assertions to af_play() (audio filtering) to make sure
input data and configured format don't mismatch. This will also catch
filters which don't set the format on their output data correctly.
Regression due to planar_audio branch.
Simulate proper handling of AOPLAY_FINAL_CHUNK. Print when underruns
occur (i.e. running out of data). Add some options that control
simulated buffer and outburst sizes.
All this is useful for debugging and self-documentation. (Note that
ao_null always was supposed to simulate an ideal AO, which is the reason
why it fools people who try to use it for benchmarking video.)
demuxer->filepos contains the byte offset of the last read packet. This
is so that the player can estimate the current playback position, if no
proper timestamps are available. Simplify it to use demux_packet->pos in
the generic demuxer code, instead of bothering every demuxer
implementation about it.
(Note that this is still a bit incorrect: it relfects the position of
the last packet read by the demuxer, not that returned to the user. But
that was already broken, and is not that trivial to fix.)
This was originally added for better seeking where libavformat's seek
function won't work well: files with timestamp resets. In these cases,
the code tried to calculate an average bitrate, and then do byte based
seeks by multiplying the seek target time with the bitrate.
Apparently this was unreliable enough that the code was just commented
(and other parts became inactive). Get rid of it.
Note that the player still does byte based seeks in these cases when
doing percent-seeks.
This should allow it to select better fallback formats, instead of
picking the first encoder sample format if ao->format is not equal to
any of the encoder sample formats.
Not sure what is supposed to happen if the encoder provides no
compatible sample format (or no sample format list at all), but in this
case ao_lavc.c still fails gracefully.
The added function af_format_conversion_score() can be used to select
the best sample format to convert to in order to reduce loss and extra
conversion work.
It calculates a "loss" score when going from one format to another, and
for each conversion that needs to be done a certain score is subtracted.
Thus, if you have to convert from one format to a set of other formats,
you can calculate the score for each conversion, and pick the one with
the highest score.
Conversion between int and float is considered the worst case. One odd
consequence is that when converting from s32 to u8 or float, u8 will be
picked.
Test program used to develop this follows:
#define MAX_FMT 200
struct entry {
const char *name;
int score;
};
static int compentry(const void *px1, const void *px2)
{
const struct entry *x1 = px1;
const struct entry *x2 = px2;
if (x1->score > x2->score)
return 1;
if (x1->score < x2->score)
return -1;
return 0;
}
int main(int argc, char *argv[])
{
for (int n = 0; af_fmtstr_table[n].name; n++) {
struct entry entry[MAX_FMT];
int entries = 0;
for (int i = 0; af_fmtstr_table[i].name; i++) {
assert(i < MAX_FMT);
entry[entries].name = af_fmtstr_table[i].name;
entry[entries].score =
af_format_conversion_score(af_fmtstr_table[i].format,
af_fmtstr_table[n].format);
entries++;
}
qsort(&entry[0], entries, sizeof(entry[0]), compentry);
for (int i = 0; i < entries; i++) {
printf("%s -> %s: %d \n", af_fmtstr_table[n].name,
entry[i].name, entry[i].score);
}
}
}
These must be written even if there was no "final frame", e.g. due to
the player being exited with "q".
Although the issue is mostly of theoretical nature, as most audio codecs
don't need the final encoding calls with NULL data. Maybe will be more
relevant in the future.
This used to be in bytes, now it's in samples. Divide the value by 8
(assuming a typical audio format, float samples with 2 channels).
Fix some editing mistake or non-sense about the extra buffering added
(1<<x instead of x<<5).
Also sneak in a s/MPlayer/mpv/.
This changes option parsing as well as filter defaults slightly. The
default is now to encode to spdif (this is way more useful than writing
raw AC3 - what was this even useful for, other than writing broken ac3
-in-wav files?). The bitrate parameter is now always in kbps.
Apparently this was completely broken after commit 22b3f522. Basically,
this locked up immediately completely while decoding the first packet.
The reason was that the buffer calculations confused bytes and number of
samples. Also, EOF reporting was broken (wrong return code).
The special-casing of ad_mpg123 and ad_spdif (with DECODE_MAX_UNIT) is a
bit annoying, but will eventually be solved in a better way.
This could cause the bundle to recache stuff because of differences with
configuration of other software using fonconfig. The defaults OS X directories
should be added to fontconfig at build time (through configure).
Slightly simplifies memory management. This might make adding a demuxer
cache wrapper easier at a later point, because you can just copy the
complete stream header, without worrying that the wrapper will free the
individual stream header fields.
The existing code tried to remove the "extra" profile flags for h264.
FF_PROFILE_H264_INTRA doesn't matter for us at all, because it's set
only for profiles the vdpau/vaapi APIs don't support.
The FF_PROFILE_H264_CONSTRAINED flag on the other hand is added to
H264_BASELINE, except that it makes the file a real subset of H264_MAIN
and H264_HIGH. Removing that flag would select the BASELINE profile,
which appears to be rarely supported by hardware decoders. This means we
accidentally rejected perfectly hardware decodable files. Use MAIN for
it instead.
(vaapi has explicit support for CONSTRAINED_BASELINE, but it seems to be
a new thing, and is not reported as supported where I tried. So don't
bother to check it, and do the same as on vdpau.)
See github issue #204.
This used to be needed for teletext support. Teletext commit has been
removed (see commit ebaaa41f), and it appears this code is inactive.
It was just forgotten with the removal. Get rid of it completely.
Untested. (Like all changes to the TV code.)
Signed-off-by: wm4 <wm4@nowhere>
Significant modifications over the original patch by not overriding
syscalls with macros ("#define open v4l2open") for fallback, but the
other way around ("#define v4l2open open"). As consequence, the calls
have to be replaced throughout the file.
Untested, although the original patch probably was tested.
Apparently this is not portable to FreeBSD. It turns out that we
(probably) don't use any symbols defined by this header directly, so
the includes are not needed.
These use the _oldargs_ hack, which failed in combination with playback
resume. Make it work.
It would be better to port all filters to new option parsing, but that's
obviously too much work, and most filters will probably be deleted and
replaced by libavfilter in the long run.
ao_null should simulate a "perfect" AO, but framestepping behaved quite
badly with it. Framstepping usually exposes problems with AOs dropping
their buffers on pause, and that's what happened here.
Most libavcodec decoders output non-interleaved audio. Add direct
support for this, and remove the hack that repacked non-interleaved
audio back to packed audio.
Remove the minlen argument from the decoder callback. Instead of
forcing every decoder to have its own decode loop to fill the buffer
until minlen is reached, leave this to the caller. So if a decoder
doesn't return enough data, it's simply called again. (In future, I
even want to change it so that decoders don't read packets directly,
but instead the caller has to pass packets to the decoders. This fits
well with this change, because now the decoder callback typically
decodes at most one packet.)
ad_mpg123.c receives some heavy refactoring. The main problem is that
it wanted to handle format changes when there was no data in the decode
output buffer yet. This sounds reasonable, but actually it would write
data into a buffer prepared for old data, since the caller doesn't know
about the format change yet. (I.e. the best place for a format change
would be _after_ writing the last sample to the output buffer.) It's
possible that this code was not perfectly sane before this commit,
and perhaps lost one frame of data after a format change, but I didn't
confirm this. Trying to fix this, I ended up rewriting the decoding
and also the probing.
libav* is generally freaking horrible, and might do bad things if the
data pointer passed to it are not aligned. One way to be sure that the
alignment is correct is allocating all pointers using av_malloc().
It's possible that this is not needed at all, though. For now it might
be better to keep this, since the mp_audio code is intended to replace
another buffer in dec_audio.c, which is currently av_malloc() allocated.
The original reason why this uses av_malloc() is apparently because
libavcodec used to directly encode into mplayer buffers, which is not
the case anymore, and thus (probably) doesn't make sense anymore.
(The commit subject uses the word "cargo cult", after all.)
Remove the awkward planarization. It had to be done because the AC3
encoder requires planar formats, but now we support them natively.
Try to simplify buffer management with mp_audio_buffer.
Improve checking for buffer overflows and out of bound writes. In
theory, these shouldn't happen due to AC3 fixed frame sizes, but being
paranoid is better.
The format negotiation is the same, except don't confusingly copy the
input format into af->data, just to overwrite it later. af->data should
alwass contain the output format, and the existing code was just a very
misguided use of the af_test_output() helper function.
Just set af->data to the output format immediately, and modify the input
format properly.
Also, if format negotiation fails (and needs another iteration), don't
initialize the libavcodec encoder.
Before this commit, the af_instance->mul/delay values were in bytes.
Using bytes is confusing for non-interleaved audio, so switch mul to
samples, and delay to seconds. For delay, seconds are more intuitive
than bytes or samples, because it's used for the latency calculation.
We also might want to replace the delay mechanism with real PTS
tracking inside the filter chain some time in the future, and PTS
will also require time-adjustments to be done in seconds.
For most filters, we just remove the redundant mul=1 initialization.
(Setting this used to be required, but not anymore.)
Since ao_openal simulates multi-channel audio by placing a bunch of
mono-sources in 3D space, non-interleaved audio is a perfect match for
it. We just have to remove the interleaving code.
ALSA supports non-interleaved audio natively using a separate API
function for writing audio. (Though you have to tell it about this on
initialization.) ALSA doesn't have separate sample formats for this,
so just pretend to negotiate the interleaved format, and assume that
all non-interleaved formats have an interleaved companion format.
Replace the code that used a single buffer with mp_audio_buffer. This
also enables non-interleaved output operation, although it's still
disabled, and no AO supports it yet.
Implementation wise, this could be much improved, such as using a
ringbuffer that doesn't require copying data all the time. This is
why we don't use mp_audio directly instead of mp_audio_buffer.
This comes with two internal AO API changes:
1. ao_driver.play now can take non-interleaved audio. For this purpose,
the data pointer is changed to void **data, where data[0] corresponds to
the pointer in the old API. Also, the len argument as well as the return
value are now in samples, not bytes. "Sample" in this context means the
unit of the smallest possible audio frame, i.e. sample_size * channels.
2. ao_driver.get_space now returns samples instead of bytes. (Similar to
the play function.)
Change all AOs to use the new API.
The AO API as exposed to the rest of the player still uses the old API.
It's emulated in ao.c. This is purely to split the commits changing all
AOs and the commits adding actual support for outputting N-I audio.
Allocate af_instance->data in generic code before filter initialization.
Every filter needs af->data (since it contains the output
configuration), so there's no reason why every filter should allocate
and free it.
Remove RESIZE_LOCAL_BUFFER(), and replace it with mp_audio_realloc_min().
Interestingly, most code becomes simpler, because the new function takes
the size in samples, and not in bytes. There are larger change in
af_scaletempo.c and af_lavcac3enc.c, because these had copied and
modified versions of the RESIZE_LOCAL_BUFFER macro/function.
No AO can handle these, so it would be a problem if they get added
later, and non-interleaved formats get accepted erroneously. Let them
gracefully fall back to other formats.
Most AOs actually would fall back, but to an unrelated formats. This is
covered by this commit too, and if possible they should pick the
interleaved variant if a non-interleaved format is requested.
Based on earlier work by Stefano Pigozzi.
There are 2 changes:
1. Instead of mp_audio.audio, mp_audio.planes[0] must be used.
2. mp_audio.len used to contain the size of the audio in bytes. Now
mp_audio.samples must be used. (Where 1 sample is the smallest unit
of audio that covers all channels.)
Also, some filters need changes to reject non-interleaved formats
properly.
Nothing uses the non-interleaved features yet, but this is needed so
that things don't just break when doing so.
This affects 64 bit floats and big endian integer PCM variants
(basically crap nobody uses). Possibly not all MS-muxed files work, but
I couldn't get or produce any samples.
Remove a bunch of format tags that are not needed anymore. Most of these
were used by demux_mov, which is long gone. Repurpose/abuse 'twos' as
mpv-internal tag for dealing with the PCM variants mentioned above.
Now to shift audio pts when outputting to e.g. avi, you need an explicit
facility to insert/remove initial samples, to avoid initial regions of
the video to be sped up/slowed down.
One such facility is the delay filter in libavfilter.
My main problem with this is that the output format will be incorrect.
(This doesn't matter right, because there are no samples output.)
This assumes all audio filters can deal with len==0 passed in for
filtering (though I wouldn't see why not).
A filter can still signal an error by returning NULL.
af_lavrresample has to be fixed, since resampling 0 samples makes
libavresample fail and return a negative error code. (Even though it's
not documented to return an error code!)
When blending OSD and subtitles onto the video, we write bogus alpha
values. This doesn't normally matter, because these values are normally
unused and discarded. But at least on Wayland, the alpha values are used
by the compositor and leads to transparent windows even with opaque
video on places where the OSD happens to use transparency.
(Also see github issue #338.)
Until now, the alpha basically contained garbage. The source factor
GL_SRC_ALPHA meant that alpha was multiplied with itself. Use GL_ONE
instead (which is why we have to use glBlendFuncSeparate()). This should
give correct results, even with video that has alpha. (Or at least it's
something close to correct, I haven't thought too hard how the
compositor will blend it, and in fact I couldn't manage to test it.)
If glBlendFuncSeparate() is not available, fall back to glBlendFunc(),
which does the same as the code did before this commit. Technically, we
support GL 1.1, but glBlendFuncSeparate is 1.4, and I guess we should
try not to crash if vo_opengl_old runs on a system with GL 1.1 drivers
only.
Apparently we were using FFmpeg-specific APIs. I have no idea whether
this code is correct on both FFmpeg and Libav (no examples, bad
doxygen... why do they even complaint aht people are using their APIs
incorrectly?), but it appears to work on FFmpeg. That was also the case
before commit ebc4ccb though, where it used internal libavformat
symbols.
Untested on Libav, Travis will tell us.
Set the PulseAudio stream title, just like the VO window title is set.
Refactor update_vo_window_title() so that we can use it for AOs too.
The ao_pulse.c bit is stolen from MPlayer.
In theory, af_volume could use separate volume levels for each channel.
But this was never used anywhere.
MPlayer implemented something similar before (svn r36498), but kept the
old path for some reason.
This member was redundant. sh_audio->sample_format indicates the sample
size already.
The TV code is a bit strange: the redundant sample size was part of the
internal TV interface. Assume it's really redundant and not something
else. The PCM decoder ignores the sample size anyway.
Also do some cosmetic changes, like merging definition and
initialization of local variables.
Remove an annoying debug mp_msg() from af_open(). It just printed the
command line parameters; if this is really needed, it could be added
to af.c instead (similar as to what vf.c does).
Helps with readability. Also remove the ctx_opt_set_* helper macros and
use av_opt_set_* directly (I think these macros were used because the
lines ended up too long, but this commit removes two indentation levels,
giving more space).
This should allow to make format negotiation much simpler, since it
takes the responsibility to compare actual input and accepted input
formats from the filters. It's also backwards compatible. Filters which
have expensive initialization still can use the old method.
I have no idea what these do, but apparently they are needed to inform
ALSA about spdif configuration. First, replace the literal constant "6"
for the AES0 parameter with the symbolic constants from the ALSA
headers (the final value is the same). Second, copy paste some funky
looking parameter setup from VLC's alsa output for setting the AES1,
AES2, AES3 parameters. (The code is actually not literally copy-pasted,
but does exactly the same.)
My small but non-zero hope is that this could make DTS-HD work, or at
least work into that direction. I can't test spdif stuff though, and
for DTS-HD not even opening the ALSA device succeeds on my system.
Using spdif with alsa requires adding magic parameters to the device
name, and the existing code tried to deal with the situation when the
user wanted to add parameters too.
Rewrite this code, in particular remove the duplicated parameter string
as preparation for the next commit. The new code is a bit stricter, e.g.
it doesn't skip spaces before and after '{' and '}'. (Just don't add
spaces.)
This accessed tons of private libavformat symbols all over the place.
Don't do this and convert all code to proper public APIs. As a
consequence, the code becomes shorter and cleaner (many things the code
tried are done by libavformat APIs).
It's probably better if all auto-inserted filters are removed when doing
an af_add operation. If they're really needed, they will be
automatically re-added.
Fix the error message. It used to be for an actual internal error, but
now it happens when format negotiation fails, e.g. when trying to use
spdif and real audio filters.
Note that the change in seek_reset is not entirely equivalent: we even
drop the remainder of buffered audio when seeking. This should be more
correct, because the whole point of the reset_ao parameter is to control
whether audio queued for output should be dropped or not.
ao_lavc.c accesses ao->buffer, which I consider internal. The access was
done in ao_lavc.c/uninit(), which tried to get the left-over audio in
order to write the last (possibly partial) audio frame. The play()
function didn't accept partial frames, because the AOPLAY_FINAL_CHUNK
flag was not correctly set, and handling it otherwise would require an
internal FIFO.
Fix this by making sure that with gapless audio (used with encoding),
the AOPLAY_FINAL_CHUNK is set only once, instead when each file ends.
Basically, move the hack in ao_lavc's uninit to uninit_player.
One thing can not be entirely correctly handled: if gapless audio is
active, we don't know really whether the AO is closed because the file
ended playing (i.e. we want to send the buffered remainder of the audio
to the AO), or whether the user is quitting the player. (The stop_play
flag is overwritten, fixing that is perhaps not worth it.) Handle this
by adding additional code to drain the AO and the buffers when playback
is quit (see play_current_file() change).
Test case: mpv avdevice://lavfi:sine=441 avdevice://lavfi:sine=441 -length 0.2267 -gapless-audio
It spams these in verbose mode. It's caused by format negotiation code
in af.c. It's for the mpv format to ffmpeg-format case, and that one is
very uninteresting. (The ffmpeg supported audio formats are practically
never extended.)
This was supposed to handle preemption better. I still think the current
state isn't very nice, since the decoder can "accidentally" call the
previous render function after preemption (instead of calling the
reloaded function), so there might be issues. But all in all, this
dummy_render function is a bit confusing, and still not entirely
correct, so it's not worth it.
Turn the sample format definitions into an enum. (The format bits are
still macros.) The native endian versions of the new definitions don't
have a NE suffix anymore, although there are still compatibility defines
since too much code uses the NE variants.
Rename the format bits for special formats to help to distinguish them
from the actual definitions, e.g. AF_FORMAT_AC3 to AF_FORMAT_S_AC3.
We found that the stretching - although it usually improves the looks of
the fonts - is incorrect.
On DVD, subtitles can cover the full area of the picture, and they have
the same pixel aspect as the movie itself.
Too bad many commercially released DVDs use bitmap fonts made with the
wrong pixel aspect (i.e. assuming 1:1) - --stretch-dvd-subs will make
these more pretty then.
The previous code used the output video's pixel aspect for stretching
purposes, breaking rendering with e.g. -vf scale in the chain. Now
subtitles are stretched using the input video's pixel aspect only,
matching the intentions of the original subtitle author.
I overlooked the fact that the ffmpeg/libav build system only supports `--cc`
and completly ignores $CC. Hopefully this makes the build times a little
faster.
Fixes#332
This removes "--hwdec=crystalhd".
I doubt anyone even tried to use this. But even if someone wants to
use it, the decoders can still be explicitly invoked with e.g.:
--vd=lavc:h264_crystalhd
The only advantage our special code provided was fallback to
software decoding. (But I'm not sure how the ffmpeg crystalhd
pseudo-decoder actually behaves.)
Removing this will allow some simplifications as soon as we don't need
vdpau_old.c anymore.
This drops autorepeated key events for a number of commands. This should
help with slow situations accidentally triggering too many repeats. (I'm
not sure if this actually happened to users - maybe.)
It's not clear whether MP_CMD_COMMAND_LIST (multiple commands on one
binding separated by ';') should be repeated, or whether it should try
to do something clever. For now, disallow autorepeat with it.
This uses vdpau OpenGL interop to convert a vdpau surface to a texture.
Note that this is a bit weak and primitive. Deinterlacing (or any other
form of vdpau postprocessing) is not supported. vo_opengl chroma scaling
and chroma sample position are not supported. Internally, the vdpau
video surfaces are converted to a RGBA surface first, because using the
video surfaces directly is too complicated. (These surfaces are always
split into separate fields, and the vo_opengl core expects progressive
frames or frames with weaved fields.)
Instead of checking for resolution and image format changes, always
fully reinit on any parameter change. Let init_video do all required
initializations, which simplifies things a little bit.
Change the gl_video/hardware decoding interop API slightly, so that
hwdec initialization gets the full image parameters.
Also make some cosmetic changes.
These formats are helpful for distinguishing surfaces with and without
alpha. Unfortunately, Libav and older version of FFmpeg don't support
them, so code will break. Fix this by treating these formats specially
on the mpv side, mapping them to RGBA on Libav, and unseting the alpha
bit in the mp_imgfmt_desc struct.
Somehow the new parser ends up much smaller. Much of it is because we
don't parse some additional information. We just skip it, instead of
parsing it and then throwing it away.
More importantly, we use the physical order of entries, instead of
trying to sort them by entry number. Each "File" entry is followed by a
number that is supposed to be the entry number, and "File1" is first.
(Should it turn out that this is really needed, an additional field
should be added to playlist_entry, and then qsort().)
Make TOOLS/matroska.pl output structs with fields sorted by name in
ebml_types.h to make the order of fields deterministic. Fix warnings in
demux_mkv.c caused by the first struct fields switching between scalar
and struct types due to non-deterministic ebml_types.h field order.
Since it's deterministic now, this shouldn't change anymore.
The warnings produced by the compilers are bogus, but we want to silence
them anyway, since this could make developers overlook legitimate
warnings.
What commits 7b52ba8, 6dd97cc, 4aae1ff were supposed to fix. An earlier
attempt sorted fields in the generated C source file, not the header
file. Hopefully this is the last commit concerning this issue...
Before the bitstream_buffers field was deprecated, you had to free it,
otherwise you would leak memory.
(Although vdpau.c uses a new API, they managed to introduce a new
deprecation this quickly. This is a complaint.)
This introduces a memory leak of 12 bytes per file on every file on some
_older_ libavcodec versions. This is minor enough that I don't care.
Video has up to 4 textures, if you include obscure formats with alpha.
This means alpha formats could always overwrite the first scaler
texture, leading to corrupted video display. This problem was recently
brought to light, when commit 571e697 started to explicitly unbind all 4
video textures, which broke rendering for non-alpha formats as well.
Fix this by reserving the correct number of texture units.
mpvcore/player/playloop.c: In function 'seek':
mpvcore/player/playloop.c:209:54: warning: declaration of 'seek' shadows a global declaration [-Wshadow]
mpvcore/player/playloop.c:209:12: warning: shadowed declaration is here [-Wshadow]
mpvcore/player/playloop.c: In function 'queue_seek':
mpvcore/player/playloop.c:360:25: warning: declaration of 'seek' shadows a global declaration [-Wshadow]
mpvcore/player/playloop.c:209:12: warning: shadowed declaration is here [-Wshadow]
Signed-off-by: wm4 <wm4@nowhere>
VA-API's OpenGL/GLX interop is pretty bad and perhaps slow (renders a
X11 pixmap into a FBO, and has to go over X11, probably involves one or
more copies), and this code serves more as an example, rather than for
serious use. On the other hand, this might be work much better than
vo_vaapi, even if slightly slower.
Most hardware decoding APIs provide some OpenGL interop. This allows
using vo_opengl, without having to read the video data back from GPU.
This requires adding a backend for each hardware decoding API. (Each
backend is an entry in gl_hwdec_vaglx[].) The backends expose video data
as a set of OpenGL textures.
Add infrastructure to support this. The next commit will add support for
VA-API.
The configure followed 5 different convetions of defines because the next guy
always wanted to introduce a new better way to uniform it[1]. For an
hypothetic feature 'hurr' you could have had:
* #define HAVE_HURR 1 / #undef HAVE_DURR
* #define HAVE_HURR / #undef HAVE_DURR
* #define CONFIG_HURR 1 / #undef CONFIG_DURR
* #define HAVE_HURR 1 / #define HAVE_DURR 0
* #define CONFIG_HURR 1 / #define CONFIG_DURR 0
All is now uniform and uses:
* #define HAVE_HURR 1
* #define HAVE_DURR 0
We like definining to 0 as opposed to `undef` bcause it can help spot typos
and is very helpful when doing big reorganizations in the code.
[1]: http://xkcd.com/927/ related
Instead of having each demuxer do it (only demux_mkv actually did...),
let generic code determine whether the file is seekable. This requires
adding exceptions to demuxers where the stream is not seekable, but the
demuxer is.
Sort-of try to improve handling of unseekable files in the player. Exit
early if the file is determined to be unseekable, instead of resetting
all decoders and then performing a pointless seek.
Add an exception to allow seeking if the file is not seekable, but the
stream cache is enabled. Print a warning in this case, because seeking
outside the cache (which we can't prevent since the demuxer is not aware
of this problem) still messes everything up.
Pointless, using stream->start_pos/end_pos instead.
demux_mf was the only place where this was used specially, but we can
rely on timestamps instead for this case.
Never check s->seek (except in init), because it'd have to check
s->flags anyway. Also, for fast skippable streams (like pipes), don't
set the bit that indicates support for seek forward.
Make sure s->end_pos is always 0 for unseekable streams. Lots of code
outside of stream.c uses this to check seeking support.
Not really cargo cult, but an unexplainable, needless difference that
just exists to annoy us.
Fixes that gcc on MinGW treats format specifiers in MSVC mode. Just why?
Why?
There are some Microsoft Windows symbols which are traditionally used by
the mplayer core, because it used to be convenient (avi was the big
format, using binary windows decoders made sense...). So these symbols
have the exact same definition as the Windows one, and if mplayer is
compiled on Windows, the symbols from windows.h are used.
This broke recently just because some files were shuffled around, and
the symbols defined in ms_hdr.h collided with windows.h ones. Since we
don't have windows binary decoders anymore, there's not the slightest
reason our symbols should have the same names. Rename them to reduce the
risk for collision, and to fix the recent regression.
Drop WAVEFORMATEXTENSIBLE, because it's mostly unused. ao_dsound defines
its own version if the windows headers don't define it, and ao_wasapi is
not available on systems where this symbol is missing.
Also reindent ms_hdr.h.
Now that matroska.pl generates struct fields in deterministic order,
this should be the last time I change this.
(gcc and clang shouldn't warn about this line of code, but since they
do, we want to workaround and silence the warning anyway.)
Newer versions of perl randomize the hash used for hashes every time
it's run; this makes the order of the fields be non-deterministic. Tack
a sort there to make it deterministic. Needed to fix (or allow fixing) a
buggy gcc warning.
mpv crashed when linked files were not found. The reason was that the
chapters array contained some uninitialized data.
I have no idea how this code works (after the merge). The old code
actually seems to remove missing chapters, while the new code just
leaves them unintiialized. Work around the crash by initializing the
chapters array (and a bunch of other things) with 0, which means the
missing chapter will be located at 00:00:00 and have no name.
There is a regression since commit af0306d.
We had some code for checking profiles earlier, which was removed in
commits 2508f38 and adfb71b. These commits mentioned that (working) hw
decoding was sometimes prevented due to profile checking, but I can't
find the samples anymore that showed this behavior. Also, I changed my
opinion, and I think checking the profiles is something that should be
done for better fallback to software decoding behavior.
The checks roughly follow VLC's vdpau profile checks, although we do
not check codec levels. (VLC's profile checks aren't necessarily
completely correct, but they're a welcome help anyway.)
Add a --vd-lavc-check-hw-profile option, which skips the profile check.
Unfortunately, we can't avoid this warning 100%, because ebml_info is
written by a Perl script. I think the script writes the struct fields in
random order (thanks Perl), so there's no way to know whether the first
struct field is a scalar or a struct.
At least {0} is always valid here, even if it shows a warning. (The
compilers are wrong, see e.g. [1].)
[1] http://gcc.gnu.org/bugzilla/show_bug.cgi?id=53119
This one really did bite me hard (see previous commit), so enable it by
default.
Fix some cases of shadowing throughout the codebase. None of these
change behavior, and all of these were correct code, and just tripped up
the warning.
E.g. "-vf scale=848:480" set the w argument twice, instead of setting w
and then h.
This was caused by accidental shadowing of a local variable.
Regression since probably 4cd143e.
As preparation for resizing the window with input commands in the
following commit.
Since there are already so many functions which somehow resize the
window, add the word "highlevel" to the name of this new function.
DVD subs (rarely) have subtitle events without end timestamp. The
duration is unknown, and they should be displayed until they're
replaced by the next event.
FFmpeg fails hard to make us aware whether duration is unknown or
actually 0, so we can't distinguish between these two cases. It fails
at this twice: AVPacket.duration is set to 0 if duration is unknown,
and AVSubtitle.end_display_time has the same issue.
Add a hack that considers all bitmap subtitles with duration==0 as
events with uknown length. I'd rather accidentally display a hidden
subtitle (if they exist at all), instead of not displaying random
subtitles at all.
See github issue #325.
The code was selecting PA_CHANNEL_POSITION_MONO for MP_SPEAKER_ID_FC,
which is correct only with the "mono" channel layout, but not anything
else. Remove the mono entry, and handle mono separately.
See github issue #326.
We mixed the "old" AVFrame management functions (avcodec_alloc_frame,
avcodec_free_frame) with reference counting. This doesn't work
correctly; you must use av_frame_alloc and av_frame_free. Of course
ffmpeg doesn't warn us about the bad usage, but will just mess up
things silently. (Thanks a lot...)
While the alloc function seems to be 100% compatible, the free function
will do bad things, such as freeing memory that might still be
referenced by another frame. I didn't experience any actual bugs, but
maybe that was pure luck.
This stopped working when the code was changed to create a window even
if --wid is used.
It appears we can't create our own window in this case, because in X11
there is no difference between a window with the root window as parent,
and a window that is managed by the WM. So make this (kind of worthless)
special case use the root window itself.
Just doing this because mp_osd.h and osd.c is not consistent.
There are some other header files (command.h and screenshot.h), but
since I don't feel too good about inflating mp_core.h, I'm not merging
them, at least not yet.
I considered making a header file for each .c file, but decided against
it. Asking around, not making separate headers was deemed acceptable. In
the end, all of these depend on MPContext and store state inside of it,
so separate headers aren't all that useful anyway.
mplayer.c was a bit too big. Split it into multiple files. I hope the
way it's split makes sense. Maybe some things don't make too much sense,
or go against intuition. These will fixed as soon as I notice them.
Some files are a bit questionable (misc.c, osd.c, configfiles.c), and
suggestions how to organize this better are welcome.
Regressions are possible due to reorganized include statements.
Obviously I didn't just copy mplayer.c's orgy of include statements, but
recreated them for each file. It's easily possible that there are
oversights and mistakes, which will show up on other platforms.
There is one actual change: the public avutil.h include is removed from
encode.h, and I tried to replace most FFMIN/FFMAX/av_clip uses. I
consider using libavutil too much as dangerous, because the set of
include files they recursively pull in is rather arbitrary and is
different between FFmpeg and Libav.
All these files access mp_core.h and MPContext, and form the actual
player application. They should be all in one place, and separate
from the other sources that are mere utility helpers.
Preparation for splitting mplayer.c into multiple smaller parts.
On systems that provide legacy OpenGL (up to 2.1), but not GL3 and
later, creating a GL3 context will fail. We then revert to legacy GL.
Apparently the error message printed when the GL3 context creation
fails is confusing. We could just silence it, but there's still a X
error ("X11 error: GLXBadFBConfig"), which would be quite hard to
filter out. For one, it would require messing with the X11 error
handler, which doesn't even carry a context pointer (for application
private data), so we don't even want to touch it. Instead, change
the error message to inform the user what's actually happening: a
fallback to an older version of OpenGL.
When starting mpv with nohup, file descriptor 0 seems to be invalid for
some reason. (I'm not quite sure why it should be... /proc/pid/fd/0
seems to indicate it's just /dev/null, and using /dev/null explicitly
shows that it works just fine.)
select() will always immediately return, and this causes mpv to burn CPU
without reason. Fix this by treating it as EOF when read() returns
EBADF.
Also add EINVAL to this condition, because it seems like a good idea.
This might actually fix an issue with DVB channel switching, because
that uses some sort of hack to reinitialize the demuxer, which causes
the subtitle renderer to initialize twice. As consequence, the assert in
add_subtitle_fonts_from_sources() would trigger.
I didn't actually test the DVB case, because I don't have the hardware.
Disable autorepeat for the "add"/"cycle" commands when changing
properties that are choices (such as fullscreen, pause, and more).
In these cases, autorepeat is not very useful, and can rather harmful
effects (like triggering too many repeated commands if command
execution is slow).
(Actually, the commands are just dropped _after_ being repeated, since
input.c itself does not know enough about the commands to decide whether
or not they should be repeated.)
Trying to toggle the border during fullscreen (with "cycle border")
would leave the window stuck without border, and it couldn't be
restored. This was because vo_x11_decoration() always excepted to be
called when toggling the state, and thus confusing the contents of the
olddecor variable. Add got_motif_hints to hopefully prevent this.
Also, when changing the border, don't take fs in account. May break on
older/broken WMs, but all in all is in fact more robust and simpler,
because you do not need to update the border state manually when
returning from fullscreen.
Defining names like min, max etc. in an often used header is not really
a good idea.
Somewhat similar to MPlayer svn commit 36491, but don't use libavutil,
because that typically causes us sorrow.
Roughly follows MPlayer svn commits 36492 and 36493. We also remove
the volume peak reporting. (There are much better libavfilter filters
for this, I think.)
Not up to my quality standards (need ASS boundingboxes), so disabled by default and undocumented for now.
Can be enabled with seektooltip=yes in plugin_osc.conf
There are certain cases where mpv will automatically set options, such
as per-file configs, per protocol/VO/AO/extension profiles, and
watch_later resume configs. All these were overwriting the user's
options, even when they were specified on command line.
Add something that explicitly preserves command line options. This
means you can now actually use the command line to override any
options that the playback resume functionality backups and restores.
It still happily overrides options set at runtime (e.g. changed via
properties while playing a file; then playing the next file might
override them again), but maybe that's not a problem with typical use.
It's true that ALSA uses alloca() in some of its API functions, but
since this is hidden behind macros in the ALSA headers, we have no
reason to include alloca.h ourselves.
Might help with portability (FreeBSD).
In cases we're just listing options or checking their values (as it
happens with -vo/-vf etc. suboption parsing), we don't need to allocate
abd initialize the actual option struct. All we're interested in is
the list of options.
The default behavior of weston changed some time ago to not fill the surface
black for fullscreen windows.
Now let mpv draw the whole screen in fullscreen mode.
Keep track of the default values directly, instead of creating a new
instance of the option struct just to get the defaults.
Also get rid of the special handling of m_obj_desc.init_options.
Instead, handle it purely by the option parser. Originally, I wanted to
handle --vo=opengl-hq and --vo=direct3d_shaders with this (by making
them aliases to the real VOs with a different preset), but since --vo
=opengl-hq=help prints the wrong values (as consequence of the
simplification), I'm not doing that, and instead use something
different.
Might reduce memory overhead, and is also less annoying.
Since pointers to m_config_options are kept around, this assumes
they're added only during initialization. Otherwise you'd get
dangling pointers.
Minor simplification. String options are now not allowed to use the
same variable/field anymore. (Also affects other "dynamic" options
which require memory allocation.)
This seems to be a problem only in OS X 10.9. I guess they improved the
general speed of the Cocoa startup and suddenly mpv core takes more time
than the Cocoa thread to initialize.
Fixes#285
(hopefully!)
Drop the author and comment fields. They were completely unused - not
even printed in verbose mode, just dead weight.
Also use designated initializers and drop redundant flags.
Set the input/output format in filter init. This doesn't change anything
functionally, but it makes the forced format show up in the filter chain
init verbose output (which sometimes prints the filter chain before all
filters have been configured).
af_format is the old audio conversion filter. It could do all possible
conversions supported by the audio chain. However, ever since the
addition of af_lavrresample, most conversions are done by
libav/swresample, and af_format is used as fallback.
Separate out the fallback cases and remove af_format. af_convert24 does
24 bit <-> 32 bit conversions, while af_convertsignendian does sign and
endian conversions. Maybe the way the conversions are split sounds a bit
odd. But the former changes the size of the audio data, while the latter
is fully in-place, so there's at least different buffer management.
This requires a quite complicated algorithm to make sure all these
"partial" conversion filters can actually get from one format to
another. E.g. s24le->s32be always requires convertsignendian and
convert24, but af.c has no idea what the intermediate format should
be. So I added a graph search (trying every possible format and
filter) to determine required format and filter. When I wrote this,
it seemed this was still better than messing everything into
af_lavrresample, but maybe this is overkill and I'll change my
opinion. For now, it seems nice to get rid of af_format though.
The AC3->IEC61937 conversion isn't supported anymore, but I don't think
this is needed anywhere. Most AOs test all formats explicitly, or use
the AF_FORMAT_IS_IEC61937() macro (which includes AC3).
One positive consequence of this change is that conversions always
include dithering (done by libav/swresample), instead of possibly going
through af_format, which doesn't do anything fancy.
Rename af_force to af_format. It's essentially compatible with command
line uses of af_format. We retain a compatibility alias for af_force.
At least not with ffmpeg.
Honestly, I have no idea how little endian AC3 works at all, since
ao_pcm doesn't do anything special about it, and treats it like s16le.
Maybe it's broken and ffmpeg has special logic to detect it.
This is obviously not needed, and just creates potential for bad
breakages (e.g. what happens if libquvi tries to open a normal filename
as http URL?).
Note that for simplicity, we still pass file:// URIs to quvi, and we
don't exclude other protocol prefixes either. In general, we don't know
what protocols quvi might support, so we don't try to second-guess it.
(Even though in practice, it's probably only "http" and "https".)
The code did not set and unset the current context inside sync sections. I am
not sure if this was an actual problem but this is better since the context is
linked to a single thread. In my brief tests this seems to avoid garbage to
show up in fullscreen.
Instead of removing dragging we now test if it we should drag the window or
not. Because if the OSC shows up we can not drag the window because that would
cause mouse events that makes the OSC disappear.
gcc and clang happen to allow {} to default-initialize a struct, but
strictly speaking, C99 requires at least {0}. In one case, we use {{0}},
but that's only because gcc as well as clang are too damn stupid not
to warn about {0}, which is a perfectly valid construct in this case.
(Sure is funny, don't warn about the non-standard case, but warn about
another standard conform case.)
Leaving these braces away just because the syntax allows them is really
obnoxious. It removes the visual cues which help understanding the code
at the first look.
For the record,
if (cond)
something();
is ok, as long as there's no else branch, and the if body is one
physical line. But everything else should have braces.
This was probably not a real problem. But it's not entirely clear
whether this could actually happen or not, so it's better to be
defensive. The code is now also somewhat easier to understand.
If cache was enabled, using ordered chapters could easily crash. The
reason is that enable_cache() reopens the demuxer and closes the old
one. The code after that (reading m->ordered_chapters etc.) then
accessed freed data.
This commit also avoids enabling cache for files which are not used
(which would make opening much slower).
Implement MP_GROW_ARRAY using MP_TARRAY_GROW. MP_GROW_ARRAY is basically
the earlier version of MP_TARRAY_GROW, and had different semantics. When
I added MP_TARRAY_GROW, I didn't dare to change it, but I think all code
that relied on the exact semantics of MP_GROW_ARRAY is gone now, or the
difference doesn't matter anyway. The semantic change is that now
(n+1)*2 elements are preallocated instead of n*2.
Also, implement MP_TARRAY_GROW using MP_RESIZE_ARRAY, which saves 1 line
of code.
In future, these macros should be part of TA.
Was disabled by default, was never used, internal support was
inconsistent and poor, and there has been virtually no interest in
creating translations.
And I don't even think that a terminal program should be translated.
This is something for (hypothetical) GUIs.
This adds support for ChapterSegmentEditionUID (pull request #258),
and also fixes issue #278 (pull request #292).
In fact, this is a straight merge of pr/292, which also contains pr/258.
When adding or removing frames to avoid 1-frame seeks on chapter
boundaries, the end of the chapter needs to be pushed or pulled by the
same amount to keep the intended end frame the same.
Note that you still need --vd-lavc-o='strict=-2' to enable the decoder.
Also, there's no guarantee that all required features for HEVC demuxing
are actually implemented, nor that the current muxing schema is the
final one.
This could cause the OSC to be displayed without mouse interaction: for
example, starting mpv with --fs, and putting the mouse to where the OSC
area is beforehand, would cause the OSC to appear and stay visible. We
don't want that. The simplest solution is not generating artificial
mouse move events from mouse enter events, because they make the OSC
think the mouse was actually moved.
Also see commit 0c7978c, where handling of mouse enter events was added.
This was supposed to fix certain corner cases, but they're not relevant
anymore due to changes in OSC behavior.
Commit 9777047 fixed this as well (by resetting the mouse state on
MOUSE_LEAVE), but all the behavior reverted with this commit as perhaps
a bad idea. It wasn't very robust, made it hard to distinguish real
events from artificial ones, and finally made the mouse cursor more
often visible than needed. (Now switching between workspaces doesn't
make the cursor visible again when switching to a fullscreened mpv.)
Now you can pretend the config file is quite literally command line
values dumped into a file, e.g.
--option1=value
--option2=value
...
although the underlying mechanisms are quite different.
Until now it used both char[] and bstr variants in the same code, which
was nasty. For example, the next commit would have additionally required
using memmove() to remove the prefix from the char[] string.
This fixes handling of e.g. "--vf=lavfi=[ noise ]" when used with
playback resume functionality. The spaces made it bug out, and there are
more cases where it could potentially break.
We could always escape for simplicity, but for now make old and new mpv
builds somewhat interoperable and use this new escaping only if needed.
This parses "%len%string" escapes, where string can contain any
characters. This method of escaping has also been used in other parts
of mplayer and mpv, so it's not a new idea.
(Also, don't confuse with URL encoding.)
Needed by the following commit.
With the way I've been doing releases in the release/0.1 branch, this
proved completely useless. You need to write the VERSION file anyway,
and since we use github's automatically generated tarballs (via its
release system), the VERSION file needs to be in the git revision of
a release anyway. git master on the other hand displayed the v0.1.0
tag, because the tag is within the master branch.
This essentially reverts commit b27f65a. Now just print the contents
of the VERSION file if it exists, and likewise, we append the git
revision if it exists. (Do that even if VERSION exists, because this
way we can tell if someone is using the release branch at an in-between
point where no new release has been made yet.)
Also change what the FFmpeg version info looks like, and additionally
dump lavfi/lavr/lswr versions. (Don't bother with libavdevice and
libpostproc, they're not important enough.)
Unfortunately, there's no "single" FFmpeg/Libav version due to fatal
braindeath on the FFmpeg/Libav side. We can't map the versions to
releases either (it simply isn't accessible anywhere).
The mouse area that covers the OSC is now only activated when the OSC is actually visible, to make sure the mouse still hides if it happens to be parked in the OSC area without making the OSC show up.
vo_image didn't handle OSD redrawing correctly anymore after OSD
redrawing behavior was changed in commit ed9295c (or maybe it has been a
problem for a longer time, and only showed up now). Basically, flip_page
was called unexpectedly and when no image was stored, which made it
crash trying to access the image. This could happen when for example
provoking OSD redrawing by pausing while using --vo=image, or by using
this command line: mpv --vo=image '-vf=lavfi="select=not(mod(n\,3))"'
Fix by removing the code that pretends vo_image can redraw OSD, and by
removing the framestepping fallback, which could make bad things happen
if the VO didn't support OSD redrawing. By now, there aren't any real
VOs that can't redraw the OSD properly, so this code is not needed and
just complicates things like vo_image.
This change likely will also be useful for vo_lavc (encoding).
Libass is technically an optional dependency, but in practice users
tend to disable libass accidentally or for the hell of it to get
something "minimal", without being aware of the consequences.
The deadzone-size is now by default zero, so movement on the entire window will make the OSC show up. To avoid it showing up by randomly moving mice, the option 'minmousemove' controls how many pixels movement (default: 1) between ticks (frames) are necessary to make the OSC show up.
The deadzone can be reenabeled by setting the option 'deadzonesize' (default: 0 = no deadzone, 1 = entire area between OSC and opposite window border), to restore the old behavior, set it to ~0.92.
The OSC will hide immediately when leaving the window or entering the deadzone (if existing) or after the time specified with 'hidetimeout' (default: 500ms) passed without any new movement. Set to negative value to disabling auto-hide (thus restoring old behavior). The OSC will never hide if hovered by the mouse.
Now that talloc has been removed, the license can be switched back to
GPLv2+. Actually, there never was a GPLv2+ licensed MPlayer (fork or
not) until now, but removal of some GPLv2-only code makes this possible
now. Rewrite the Copyright file to explain the reasons for the licenses
MPlayer and forks use. The old Copyright file didn't contain anything
interesting anymore, and all information it contained is available at
other places in the source tree.
The reason for the license change itself is that it should improve
interoperability with differently licensed code in general.
This essentially reverts commit 1752808.
There are multiple reasons to do this. One big reason is the license:
talloc is LGPLv3+, which forces mpv to be licensed as GPLv3+.
Another one is that our talloc copy contains modifications, which makes
it essentially incompatible with upstream talloc (in particular, our
version aborts on out of memory conditions - well, it wasn't my idea).
Updating from upstream is also a bit involved - the talloc source is
not really organized in a way to allow copying it into projects (and
this isn't an intended use-case).
Finally, talloc is kind of big and bloated. The replacement halves the
amount of code - mainly because we didn't use all talloc features. It's
even more extreme if you compare upstream talloc (~4700 lines) and the
new allocator without talloc compat (~900 lines).
The replacement provides all features we need. It also doesn't clash
with talloc. (The talloc compatibility wrapper uses macros to avoid
introducing linker-level symbols which could clash with libtalloc.)
It also tries to lower the overhead (only 4 words opposed to 10 words
in talloc for leaf nodes in release mode). Debugging features like leak
reporting can be enabled at compile time and add somewhat more overhead.
Though I'm not sure whether the overhead reduction was actually
successful: allocations with children need an "extra" header, which adds
plenty of overhead, and it turns out that almost half of all allocations
have children. Maybe the implementation could be simplified and the
extra header removed - even then, overhead would be lower than talloc's.
Currently, debugging features can be entirely deactivated by defining
NDEBUG - I'm not sure if anything defines this directly yet, though.
Unlike in talloc, the leak reporting stuff is thread-safe. (That's also
why it's far less elegant, and requires extra list pointers.)
Comes with a compatibility layer, so no changes to mpv source code
are needed. The idea is that we will pretend to be using talloc for
a while, so that we can revert to our old talloc implementation at
any time for debugging purposes.
Some inspiration was taken from Mesa's ralloc:
http://cgit.freedesktop.org/mesa/mesa/tree/src/glsl/ralloc.h
This is another talloc replacement, but lacks some features we need
(getting size of an allocation, debugging features, being able to
access children in the dtor).
There's some information in ta/README what will happen next and how the
transition is expected to progress.
Change talloc destructor so that they can never signal failure, and
don't return a status code. This makes our talloc copy even more
incompatible to upstream talloc, but on the other hand this is
preparation for getting rid of talloc entirely.
(The talloc replacement in the next commit won't allow the talloc_free
equivalent to fail, and the destructor return value would be useless.
But I don't want to change any mpv code either; the idea is that the
talloc replacement commit can be reverted for some time in order to
test whether the talloc replacement introduced a regression.)
This only shows any differences when mpv isn't frontmost and is in fullscreen.
Cmd+Tab overlay is still at a higher level as to avoid complete usability fail.
Before this, they were shown on terminal only. Now they use the OSD
mechanism, which shows them on the video window or the terminal,
depending on settings and what's available.
Changing volume when audio is disabled was a feature request (github
issue #215), and was introduced with commit 327a779.
But trying to fix github issue #280 (volume is not correct in no-audio
mode, and if audio is re-enabled, the volume set in no-audio mode isn't
set), I concluded that it's not worth the trouble and the current
implementation is questionable all around. (For example, you can't
change the real volume in no-audio mode, even if the AO is open - this
could happen with gapless audio.) It's hard to get right, and the
current mixer code is already hilariously overcomplicated. (Virtually
all of mixer.c is an amalgamation of various obscure corner cases.)
So just remove this feature again.
Note that "options/volume" and "options/mute" still can be used in
idle mode to adjust the volume used next time, though these properties
can't be used during playback and thus not in audio-only mode.
Querying the volume still "works" in audio-only mode, though it can
return bogus values.
The argument or this change is that --loop should set how often the
file is played, not the number of additional repeats.
Based on pull request 277, with additions to the manpage and removal
of "--loop=0".
Signed-off-by: wm4 <wm4@nowhere>
The primary effect of this commit is that it fixes playback resume if
libquvi 0.9 and German locale are used, and libkdecore is on the system.
(See github issue #279.)
libquvi uses libproxy to determine system-wide proxy settings. libproxy
in turn loads libkdecore (if present) to determine KDE proxy settings.
libkdecore has a global constructor, which calls setlocale(LC_ALL, ""),
switching the current locale from "C" to what the user's settings. This
breaks some basic C string processing functions. Note that the locale
won't be set on program start, but only when libproxy calls dlopen() on
its config_kde module, which actually causes libkdecore to be loaded and
initialized.
In particular, with German locale, snprintf() would use "," instead of
"." when formatting float values, which in combination with playback
resume, would lead to parse errors the next time mpv was started, which
is how this issue was found.
I'd consider this a bug with libkdecore or at least libproxy. No library
should ever even touch locale: it might break basic expectations on C
string handling functions, might override program settings, and it's not
thread-safe. But this is so nasty and severe, that a quick hack to fix
it hurts less.
See github issue #279 and KDE bug #325902.
Before this patch, those will cause a crash:
mpv -playlist /dev/null
mpv -playlist <(bla) # if the result of bla is empty
Signed-off-by: Mohammad Alsaleh <CE.Mohammad.AlSaleh@gmail.com>
Signed-off-by: Stefano Pigozzi <stefano.pigozzi@gmail.com>
previously, mp_ring_read_cb() would allow reading past the end of the
ringbuffer when buffered > available. mp_ring_read() had logic for
splitting the read into two, which I duplicated into mp_ring_read_cb().
Since ordered editions can reference external files, when an ordered
chapter references another ordered edition, we have to go and figure out
what is is referencing as well.
When playing a Matroska file with ordered chapters, it may reference
another file by edition uid. When this edition is also ordered, it may
reference other files. When this occurs, the new segment/edition pair is
added to the list of sources to search for.
To support edition references in matroska chapters, editions need to be
remembered for each chapter and source. To facilitate easier management
of these now-paired uids, a single structure is used.
First, don't try to seek if the result is 0 (i.e. nothing found, or
subtitle event happens to be exactly on spot).
Second, since we never can make sure that we actually seek to the exact
subtitle PTS (seeking "snaps" to video PTS), offset the seek by 10ms.
Since most subtitle events are longer than 10ms, this should work fine.
The OSC will now display cache fill status between the timecodes, but only if it's below 48% to not clutter the interface with erratically changing values.
By default, the displayed value is multiplied by 2 to not confuse users who are unfamillar with the inner workings of the caching system. This can be disabled using the iAmAProgrammer=true setting.
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@36461 b3059339-0415-0410-9bf9-f77b7e298cf2
Fixes playback of http://mpg123.org/test/44and22.mp3
Cherry-picked from MPlayer SVN rev. #36461, a patch by
Thomas Orgis, committed by by Reimar Döffinger.
You couldn't jump to the first entry in the playlist, if that entry was
marked as "too short". But this is usually fine, because it doesn't get
into the way of the user. (This feature is meant to allow being able to
jump backwards through the playlist even if a file immediately stops
playback right after initialization, not to prevent playing these files
at all.)
Also, apply the skip logic when wrapping around the playlist when going
backwards. This happens with --loop, because looping pretends the
playlist is infinitely repeated forwards and backwards (as long as the
playlist_prev/next commands are concerned).
Also add some comments.
When for example switching off the video stream, and --force-window is
used, forcefully reconfigure the VO. This will reset the size, and will
clear the window with black.
Needed some effort, because you don't always want to clear the window
on "discontinuity" points like going to a next file: this would
introduce some flicker.
glXGetVisualFromFBConfig() specifies specifies that it can return NULL
if there is no associated X visual. Instead of crashing, let
initialization fail. I'm not sure if this is actually supposed to work
with a fallback visual (passing a NULL visual to vo_x11_config_vo_window
would just do this), but let's play safe for now.
Apparently this can happen when trying to use vo_opengl over a remote
X display.
Output silence to the output buffer during underruns. This removes small
occasional glitches that happen before the AUHAL is actually paused from the
`audio_pause` call.
Fixes#269
Reverts a small change made in commit ed9295c. This is needed, because
otherwise mplayer.c/update_video_attached_pic() thinks it never has to
update the picture after initialization. (Maybe there would be more
elegant ways to handle this, but not without adding extra state.)
Didn't handle VO events, didn't handle OSD message management.
There is probably still some strangeness left. --idle mode was never
made for direct interaction.
We can render subtitles if a VO is open. Whether we're decoding video
(i.e. if mpctx->sh_video is set) doesn't really matter.
Subtitle display with --force-window still doesn't quite work yet,
because there's nothing to actually force redrawing of subtitles in this
mode.
This commit adds the --force-window option, which will cause mpv always
to create a window when started. This can be useful when pretending that
mpv is a GUI application (which it isn't, but users pretend anyway), and
playing audio files would run mpv in the background without giving a
window to control it.
This doesn't actually create the window immediately: it only does so
only after initializing playback and when it is clear that there won't
be any actual video. This could be a problem when starting slow or
completely stuck network streams (mpv would remain frozen in the
background), or if video initialization somehow is stuck forever in
an in-between state (like when the decoder doesn't output a video
frame, but doesn't return an error either). Well, we can pretend only
so much that mpv is a GUI application.
Seeking normally resets the VO and throws away the currently displayed
frame, so if you seek outside the video with --keep-open enabled, the
window would normally be "stuck" and not redraw properly, because there
is no source video frame that could be redrawn. To deal with this, a
precise seek to the position of the last displayed video frame was
issued.
This usually worked, but it can fail for formats where seeking is broken
or not possible (consider reading from a pipe).
Fix this by changing the semantics for vo_seek_reset(): now the video
frame is remembered even after seeking.
Note that this changes behavior a little when trying to seek outside of
a file with --keep-open enabled. Since no actual seek is done anymore,
the video will remain "frozen" on the previous position, and you can't
unpause or framestep to see the video between current position and
actual end of the video. If users complain, I might revert this commit.
vo_vdpau is the only VO which implements VOCTRL_RESET. Redrawing the
last output frame is hard, because the output could consist of several
source video frames with certain types of post-processing
(deinterlacing). Implement redrawing as special case by keeping the
previous video frames aside until at least one new frame is decoded.
This improves the previous commit, but is separate, because it's rather
complicated.
Before, a VO could easily refuse to respond to VOCTRL_REDRAW_FRAME,
which means the VO wouldn't redraw OSD and window contents, and the
player would appear frozen to the user. This was a bit stupid, and makes
dealing with some corner cases much harder (think of --keep-open, which
was hard to implement, because the VO gets into this state if there are
no new video frames after a seek reset).
Change this, and require VOs to always react to VOCTRL_REDRAW_FRAME.
There are two aspects of this: First, behavior after a (successful)
vo_reconfig() call, but before any video frame has been displayed.
Second, behavior after a vo_seek_reset().
For the first issue, we define that sending VOCTRL_REDRAW_FRAME after
vo_reconfig() should clear the window with black. This requires minor
changes to some VOs. In particular vaapi makes this horribly
complicated, because OSD rendering is bound to a video surface. We
create a black dummy surface for this purpose.
The second issue is much simpler and works already with most VOs: they
simply redraw whatever has been uploaded previously. The exception is
vdpau, which has a complicated mechanism to track and filter video
frames. The state associated with this mechanism is completely cleared
with vo_seek_reset(), so implementing this to work as expected is not
trivial. For now, we just clear the window with black.
Trying to connect multiple mpv clients to JACK with the
JackUseExactName option would fail unless the user manually
specifies a unique client name. This changes the behavior
to automatically generate a unique name if the requested
one is already in use.
vo_x11 had a clever trick to implement a video equalizer: it requested a
DirectColor visual. This is a X11 mechanism which allows you to specify
a lookup table for each color channel. Effectively, this is a safe
override for the graphic card's gamma ramp. If X thinks the window
deserves priority over other windows in the system, X would temporarily
switch the gamma ramp so that DirectColor visuals can be displayed as
the application intends. (I'm not sure what the exact policy is, but in
practice, this meant the equalizer worked when the mouse button was
inside the window.)
But all in all, this is just lots of useless code for a feature that is
rarely ever useful. Remove it and use the libswscale equalizer instead.
(This comes without a cost, since vo_x11 already uses libswscale.)
One worry was that using DirectColor could have made it work better in
8-bit paletted mode. But this is not the case: there's no difference,
and in both cases, the video looks equally bad.
Considering that we use version control, maintaining an AUTHORS file
doesn't make much sense, and indeed it was barely touched since the time
mplayer2 forked from MPlayer. None of the contributors to mplayer2 or
mpv are listed in this file (even uau's work is not appropriately
reflected, despite having an entry in this file).
I'm removing this file because it's obviously pointless to keep around,
and moreover it's misleading.
In theory removing this file might be questionable: the git log reaches
back into CVS and SVN times, which don't have separate committer and
author fields for each comment. The commit messages could be incomplete
and not mention the original author, while the AUTHORS file somehow
could. I doubt this is a real concern, since you can always reconstruct
the original AUTHORS file from git.
Another concern is that this could be interpreted as disrespectful to
the people mentioned in this file; but it isn't meant to be
disrespectful.
Also point out a slight irregularity in the Copyright file.
Commit 884c179 attempted to make it possible to skip backwards through
the playlist, even for files which fail to intitialize, or play for a
very short time. This was also used to prevent mpv from looping forever
and doing nothing if --loop=inf is used, and no file in the playlist is
playable.
This broke looping of very short files, because mpv was assuming that
this case happened. But there are legitimate use cases.
Fix this by making the looping case special. Instead of checking whether
playback was "very short", check whether something could be decoded and
displayed/played. If yes, allow looping.
After rebasing my dev branch it turned out that the code deadlocked on
recursive calls of `vo_control`. Make the locking code a little bit smarter
by making always skip locking/unlocking if we are executing a chunck of
code that is already synchronized with `dispatch_sync`.
Split the code to several files. The GUI elements now each have they own files
and private state. The original code was a mess to respect the retarded mplayer
convention of having everything in a single file.
This commit also seems to fix the long running bug of artifacts showing
randomly when going fullscreen using nVidia GPUs.
Don't allocate a VAImage and a mp_image every time. VAImage are cached
in the surfaces themselves, and for mp_image an explicit pool is
created. The retry loop runs only once for each surface now.
This also makes use of vaDeriveImage() if possible.
Until now, mouse positions were just passed to the core as-is, even if
the mouse coordinates didn't map to any useful coordinate space, like
OSD coordinates. Lua scripting (used by the OSC, the only current user
of mouse input) had to translate mouse coordinates manually to OSD space
using mp_get_osd_mouse_pos(). This actually didn't work correctly in
cases mouse coordinates didn't map to OSD (like vo_xv): the mouse
coordinates the OSC got were correct, but input.c was still expecting
"real" mosue coordinates for mouse areas.
Fix this by converting to OSD coordinates before passing the mouse
position to the core.
Do this so that MOUSE_LEAVE can't be combined with other keys. (E.g.
keep 'w' pressed, then move the mouse outside of the mpv window; it will
print a warning what w-MOUSE_LEAVE is not mapped.)
It appears the last run missed all mp_tmsg().
Since the command parser also prints messages, it needs a new parameter
to pass a log context. We go the easy way and just require the input_ctx
to be passed, and use its mp_log.
There is uninitialized memory access if the actual size isn't passed
along. In the worst case, this can cause a source to be loaded against
the uninitialized memory, causing a false count of found versus required
sources, preventing the "Failed to find ordered chapter part" message.
This is pretty much a hack for the OSC. It will allow it to rely on a
somewhat predictable style, instead of having to overwrite all user
OSD settings manually with override tags.
Nothing really accesses it. Subtitle initialization actually does in a
somewhat meaningful way, but there container size is probably fine, as
subtitles were always initialized before the first video frame was
decoded.
Now writing -1 to the 'aspect' property resets the video to the auto
aspect ratio. Returning the aspect from the property becomes a bit more
complicated, because we still try to return the container aspect ratio
if no frame has been decoded yet.
This function would probably be useful in other places too.
I'm not sure why vd.c doesn't apply the aspect if it changes size by
less than 4 pixels. Maybe it's supposed to avoid ugly results with bad
scalers if the difference is too small to be noticed normally.
This is preliminary. There are still tons of issues, and any aspect
of scripting may change in the future. I decided to merge this
(preliminary) work now because it makes it easier to develop it, not
because it's done. lua.rst is clear enough about it (plus some
sarcasm).
This requires linking to Lua. Lua has no official pkg-config file, but
there are distribution specific .pc files, all with different names.
Adding a non-pkg-config based configure test was considered, but we'd
rather not.
One major complication is that libquvi links against Lua too, and if
the Lua version is different from mpv's, you will get a crash as soon
as libquvi uses Lua. (libquvi by design always runs when a file is
opened.) I would consider this the problem of distros and whoever
builds mpv, but to make things easier for users, we add a terrible
runtime test to the configure script, which probes whether libquvi
will crash. This is disabled when cross-compiling, but in that case
we hope the user knows what he is doing.
This time it broke because I didn't actually test compiling vo_vaapi.c,
and it was using a macro from mp_image.h, which implicitly assumed
FFALIGN was available. Screw that too, and copy the definition of
ffmpeg's FFALIGN to MP_ALIGN_UP, and move these macros to mp_comnon.h.
The code using FFSWAP was moved from vo_vaapi.c to vaapi.c, which didn't
include libavutil/common.h anymore, just libavutil/avutil.h. The header
avutil.h doesn't include common.h recursively in Libav, so it broke
there.
Add FFSWAP as MPSWAP in mp_common.h (copy pasted from ffmpeg) to make
sure this doesn't happen again. (This kind of stuff happens all too
often, so screw libavutil.)
This code is actually quite inefficient: it reuses the (slow, simple)
screenshot code. It uses an inefficient method to read the image
(vaGetImage() instead of vaDeriveImage()), allocates new memory for
each frame that is read, and it tries all image formats again each
time.
Also, in my tests it always picked NV12 as image format, which is not
ideal if you actually want to filter the video, and vo_xv can't handle
this format without conversion either.
However, a user confirmed that it worked for him, so everything is fine.
This will allow GPU read-back with process_image.
We have to restructure how init_vo() works. Instead of initializing the
VO before process_image is called, rename init_vo() to
update_image_params(), and let it update the params only. Then we really
initialize the VO after process_image.
As a consequence of these changes, already decoded hw frames are
correctly unreferenced if creation of the filter chain fails. This
could trigger assertions on VO uninitialization, because it's not
allowed to reference hw frames past VO lifetime.
Merged from pull request #246 by xylosper. Minor cosmetic changes, some
adjustments (compatibility with older libva versions), and manpage
additions by wm4.
Signed-off-by: wm4 <wm4@nowhere>
This code was made inactive some months ago. At this time it wasn't
entirely clear whether this code was still needed, but now I'm pretty
sure it isn't. Even if it is, it didn't work anymore.
This wasn't enough and could lead to a cut off message shown on OSD.
Just make it dynamic, since we already use dynamic memory allocation
at this point anyway.
Moving the window was convenient but generates a MOUSE_LEAVE event
which it shouldn't. Now we remove it, because it is still possible
to move the window in weston with MOD+BTN0.
The problem with DVD/BD and playback resume is that most often, the
filename is just "dvd://", while the actual path to the DVD disk image
is given with --dvd-device. But playback resume works on the filename
only.
Add a pretty bad hack that includes the path to the disk image if the
filename starts with dvd://, and the same for BD respectively. (It's a
bad hack, but I want to go to bed, so here we go. I might revert or
improve it later, depending on user feedback.)
We have to cleanup the global variable mess around the dvd_device.
Ideally, this should go into MPOpts, but it isn't yet. Make the code
paths in mplayer.c take MPOpts anyway.
By default, libavformat uses UDP for rtsp playback. This doesn't work
very well. Apparently the reason is that the buffer sizes libavformat
chooses for UDP are way too small, and switching to TCP gets rid of this
issue entirely (thanks go to Reimar Döffinger for figuring this out).
In theory, you can set buffer sizes as libavformat options, but that
doesn't seem to help.
Add an option to select the rtsp transport, and make TCP the default.
Also remove an outdated comment from stream.c.
Mainly for debugging. Usually, we just set options for all possible
protocols, and we can't really know whether a certain protocol is used
beforehand. That's also the reason why avio_open2() takes a dictionary,
instead of letting the user set options directly with av_opt_set(). Or
in other words, we don't know whether an option that could be set is an
error or not, thus we print the messages only at verbose level.
This is for properties that normally show a bar, and thus do not show an
OSD message (as per classic mplayer behavior). Setting an extra_msg
allows showing an OSD message anyway, except if OSD messages are
explicitly suppressed.
This refactors the whole show_property_osd() function a bit, and
replaces the weird sep field with a more general method.
Allows for example: --status-msg='${?pause==yes:(Paused) } ...' to
emulate the normal terminal status line. It's useful in other situations
too.
I'm a bit worried about extending this mini-DSL, and sure hope nobody
will implement a generic formula evaluator at some point in the future.
But for now we're probably safe.
In most cases, it's better if deinterlacing happens before any other
filtering, so prepend the filter to the user's filter list, instead
of appending it.
Instead of hardcoding a single filter. This might be helpful for
modeling the vaapi deinterlacer as a video filter. The idea is that a
software deinterlacer would be tried first, and if that fails (because
vaapi hardware decoding uses HW surfaces, which a software deinterlacer
does not accept), the vaapi filter would be tried.
Normally, we need this for Xutf8LookupString(). But we can just fall
back to XLookupString(). In fact, the code for this was already there,
the code was just never tested and was actually crashing when active
(see commit 2115c4a).
Calling them separately doesn't really make sense, and all existing
calls to them usually combined them. One subtitle difference was that
af_init() didn't wipe the filter chain if initialization of the chain
itself failed, but that didn't really make sense anyway.
Also remove af_init() from the code for setting balance in mixer.c. The
mixer should be in the initialized state only if audio is fully
initialized, so the af_init() call made no sense.
Note that the filter "editing" code in command.c doesn't really do a
nice job of handling errors in case recreating an _old_ (known to work)
filter chain unexpectedly fails, and this obscure/rare case might be
differently handled after this change.
The volume is set as soon as the audio chain is created again. This
works only in softvol mode. For system wide volume or otherwise
externally user controllable volume, this code is intentionally
disabled. It would be extremely weird if changing volume (while audio is
not initialized) would do nothing, and then suddenly change it when the
audio chain is created.
There's another odd corner case: the user-set volume will be thrown away
if it's set before the _first_ audio chain initialization. This is
because the volume restore logic recognizes a change from nothing to
softvol or an AO, and circumventing that would require additional code.
Also, we don't even know the start volume before that point.
Forcing the volume with --volume will can override the volume set during
no-audio mode, depending on the situation.
Note that this is intentionally never done if the AO or softvolume is
different, or if the current volume control method is thought to control
system wide volume (such as ALSA) or otherwise user controllable (such
as PulseAudio). The intention is to keep things robust and to avoid
messing with the user's audio settings as far as possible, while still
providing the ability to resume volume if it makes sense.
Commit broke text subtitles without embedded fonts. Will look for a better
solution later. Revert it for now, since I'm starting to get bug reports.
This reverts commit 4a9f618d9f.
XOpenIM can fail to find a valid input method, in which case it
returns NULL. Passing a NULL pointer to XCreateIC would cause a
crash, so fail VO init before that happens.
Before this commit there was just an error message, but the file descriptor was
still open. Now we close the file descriptor and prevent it from calling
endlessly. Also a CLOSE_WIN event is sent which closes the window eventually if
the action of CLOSE_WIN is set to quit or quit_watch_later.
Improves display of images and video with alpha channel, especially if
the transparent regions contain (supposed to be invisible) garbage
color values.
Refactor how mixer.c does volume/mute restoration and initialization.
Move to handling of --volume and --mute to mixer.c. Simplify the
implementation of these and hopefully fix bugs/strange behavior related
to using them as file-local options (this uses a somewhat dirty trick:
the option values are reverted to "auto" after initialization). Put most
code related to initialization and volume restoring in probe_softvol()
and restore_volume(). Having this code all in one place is less
confusing.
Instead of trying to detect whether to use softvol at runtime, detect it
at initialization time using AOCONTROL_GET_VOLUME (same with mute,
AOCONTROL_GET_MUTE). This implies we expect SET_VOLUME/SET_MUTE to work
if the GET variants work. Hopefully this is always the case.
This is also preparation for being able to change volume/mute settings
if audio is disabled, and for allowing restoring value with playback
resume.
Softvol always used a linear multiplier for volume control. This was
converted to dB, and then back to linear in af_volume. Remove this non-
sense. We still try to keep the command line argument to af_volume in
dB, though.
This is to avoid the 30s hang while mpv caches fonts. In practice all the
fonts an average user is going to use are embedded in mkv files so there is
no reason to build fontconfig's cache on all of OS X system directories.
I might add something similar for terminal usage, but I am highly undecided.
It's quite unlikely, but functions like mp_find_user_config_file() can
return NULL, e.g. if $HOME is unset.
Fix all the code that didn't check for this correctly yet.
Remove the ifdef hell from mp_find_user_config_file(). Move the win32
specific code (for MinGW and Cygwin) to path-win.c. The behavior should
be about the same, but I can't be sure due to lack of testing and
because the old path.c code was hard to follow. (I expect those who care
about windows will fix things, should issues pop up - sorry.)
One difference is that the new code will always force MPV_HOME. It looks
like the old code preferred the mpv config dir in the exe dir if it
exists.
Also, make sure MP_PATH_MAX has enough space, even if the equivalent
wchar_t string is not 0-terminated with PATH_MAX (because apparently the
winapi doesn't require this). (Actually, maybe we should just kill all
uses of PATH_MAX/MP_PATH_MAX.)
This is a bit "hard", because getenv() returns a static string, and we
can't just return an allocated string. We also want getenv() to be
thread-safe if possible. (If the mpv core is going to be more threaded,
we sure do want the lower layers to be thread-safe as well.)
The homepath variable was static, and its value was set to a stack
buffer. This means a second invocation of the function would trigger
undefined behavior. Moreover the stack buffer always went out of scope
before homepath was used.
This is mainly to avoid spurious cursor states due to the mouse moving inside
or outside the window as a result of the window resize (with cmd-0/1/2).
This avoids complex logic and triggers a mouse move so that the player
recomputes the correct cursor state based on the autohide configuration of
the user.
This keeps the state in sync with the current state in cocoa_common. Infact the
cocoa code in mpv can decide wether it really wants to hide the cursor based on
the result of the `canHideCursor` method (this is so that the cursor is only
hidden when hovering on the video window).
Turns out that these checks were for versions of OS X that mpv doesn't even
support anymore. So just remove the checks since they cause a deprecation
warning.
GetCurrentProcess() is deprecated on 10.9. Make a universal solution by
checking OS version number.
get_system_version() function is the recommended Apple way of getting the
OS version, since Gestalt is also deprecated (and does pretty much the same
thing anyway)
Updating HIDRemote.m to use a similar function would allow to get rid of the
2 other warnings.
This is supposed to reduce the amount of useless error messages shown
during initialization of vo_opengl. If multiple backends are compiled,
usually only one of them will work. For example, on Linux both X and
Wayland backends can be compiled, but usually either Wayland or X is
running. Then, if Wayland is not running, but X is, trying to initialize
the Wayland backend should not spam the terminal with error messages.
Signed-off-by: Andreas Sinz <andreas.sinz@aon.at>
Call update_subtitles() on every iteration of the playloop, so that
subtitle packets are read as soon as possible, instead of every time a
video frame is displayed. This helps in case the packet queue is swamped
with subtitle packets, which can happen with certain insane mkv files.
The change will simply cause the subtitle queue to be emptied on each
playloop iteration.
The timestamps update_subtitles() uses for display are the same before
and after this commit. (Important for files which have subtitle packets
with timestamps or duration not set.)
In insane files with a very huge number of subtitle events, and if the
--demuxer-mkv-subtitle-preroll option is given, seeking can still
overflow the packet queue. Normally, the subtitle_preroll variable
specifies the maximum number of packets that can be added. But once this
number is reached, the normal seeking behavior is enabled, which will
add all subtitle packets with the right timestamps to the packet queue.
At this point the next video keyframe can still be quite far away, with
enough subtitle packets on the way to overflow the packet queue.
Fix this by always setting an upper limit of subtitle packets read
during seeking. This should provide additional robustness even if the
preroll option is not used.
This means that even with normal seeking, at most 500 subtitle packets
are demuxed. Packets after that are discarded.
One slightly questionable aspect of this commit is that subtitle_preroll
is never reset in audio-only mode, but that is probably ok.
The mpv.desktop file is taken from the Arch package [1]. It appears to
be based on the mplayer2 git mplayer.desktop file (e.g. very similar
MimeType field), with minor modifications applied by Arch package
maintainers.
Note that for now, this doesn't show a terminal (Terminal=false), which
might not always be ideal. For example, if the file is audio only, or
if VO initialization fails for some reason, mpv will run in the
background and play audio without showing a window. But users prefer
running it without terminal, and don't want to play audio files with
it.
Maybe a --force-window option will be added in the future, which would
always create a VO window, and compensate for these issues.
[1] https://aur.archlinux.org/packages/mpv-git/
These would have to be updated manually all the time. Replacing them
automatically would be possible, but additional work, and would force
regeneration of the manpage way too often.
We decided that we don't need these fields.
Instead of containing a format string within %w{...}, simply allow %w
to specify one item of a time format string. This is simpler, more like
other format specifiers (%t), and probably easier to use too.
This is for situations when repeated attempts at playing a playlist
entry failed, and playlist navigation becomes impossible due to that.
For example, it wasn't possible to skip backwards past an unplayable
playlist entry:
mpv file1.mkv doesntexist.mkv file3.mkv
You couldn't skip back to file1.mkv from file3.mkv. When running a
single "playlist_prev" command, doesntexist.mkv would be played, which
would fail to load. As reaction to the failure to load it, the next file
would be played, which is file3.mkv.
To make this even worse, the file could successfully load, but run only
for a split second. So just loading successfully isn't good enough.
Attempt to solve this by marking problematic playlist entries as failed,
and by having playlist_prev skip past such playlist entries. We define
failure as not being able to play more than 3 seconds (or failing to
initialize to begin with). (The 3 seconds are in real time, not file
duration.)
"playlist_prev force" still exhibits the old behavior.
Additionally, use the same mechanism to prevent pointless infinite
reloading if none of the files on the playlist exist. (See github issue
All in all, this is a heuristic, and later adjustments might be
necessary.
Note: forward skips (playlist_next) are not affected at all. (Except for
the interaction with --loop.)
This is commonly used to disable the screensaver with broken/non-
standard X screensavers. During pause, the screensaver should not be
disabled, so not calling this command while paused seems sensible.
See github issue #236.
They are rarely useful in my opinion.
This commit was mainly motivated by this message:
Video uses a non-standard and wasteful way to store B-frames ('packed B-frames'). Consider using a tool like VirtualDub or avidemux to fix it.
It's what's left over from the "Invalid and inefficient vfw-avi..."
warning that used to be printed when playing avi/divx files. Although
the new message is much better, it's still rather useless and poses
more questions than it answers. Besides, nobody wants to remux a file
when playing it, especially not if playback appears to be completely
fine. (There are some claims that these files raise CPU usage, but even
my old crappy CPU can decode low res avi/divx files at real time at
about x35 playback speed.)
I have a sample where some final chapters are missing. This was causing a
segmentation fault when trying to fetch chapter times for them.
This makes the code ignore those chapters.
The quicktime html scripting guide suggests to wrap urls not
necesarly associated with quicktime in a .mov file.
(so that when <embed>ing videos quicktime would be forced.)
These mov files may contain several "Text Hacks".
One of these is RTSPtext.
The suggested/allowed format (as regex) is like:
RTSPtext[ \r]RTSP://url
See also p.51 of:
https://developer.apple.com/library/mac/documentation/QuickTime/Conceptual/QTScripting_HTML/QTScripting_HTML.pdf
In reality there are also files like (e.g. zdfmediathek.de):
RTSPtext\nrtsp://url\n\n
Lets handle these files as a playlist with one element.
In init_vo(), if sh->aspect is 0 or last_sample_aspect_ratio is set,
sh->aspect is overwritten. With software decoding fallback behaviour,
this makes the aspect ratio from container ignored since
last_sample_aspect_ratio is already set in first try with hardware
decoding.
The --deinterlace option does on playback start what the "deinterlace"
property normally does at runtime. You could do this before by using the
--vf option or by messing with the vo_vdpau default options, but this
new option is supposed to be a "foolproof" way.
The main motivation for adding this is so that the deinterlace property
can be restored when using the video resume functionality
(quit_watch_later command).
Implementation-wise, this is a bit messy. The video chain is rebuilt in
mpcodecs_reconfig_vo(), where we don't have access to MPContext, so the
usual mechanism for enabling deinterlacing can't be used. Further,
mpcodecs_reconfig_vo() is called by the video decoder, which doesn't
have access to MPContext either. Moving this call to mplayer.c isn't
currently possible either (see below). So we just do this before frames
are filtered, which potentially means setting the deinterlacing every
frame. Fortunately, setting deinterlacing is stable and idempotent, so
this is hopefully not a problem. We also add a counter that is
incremented on each reconfig to reduce the amount of additional work per
frame to nearly zero.
The reason we can't move mpcodecs_reconfig_vo() to mplayer.c is because
of hardware decoding: we need to check whether the video chain works
before we decide that we can use hardware decoding. Changing it so that
this can be decided in advance without building a filter chain sounds
like a good idea and should be done, but we aren't there yet.
I did commit 86c05655d by thinking `mpv` already removed the `mpv` from
argc/argv. It actually is still there, so the argc must be 1 to check for no
arguments.
Thanks to @Nyx0uf for pointing out the bug and for testing on 10.9!
File opening through Finder, apparently drops `--psn` arguments on Mavericks
and just uses no args. Modify the code to account for that case.
This wasn't tested on 10.9 itself (I don't have a paid dev account), but it
*should* work if I understood the problem correctly.
Problem: I own the buffer and I destroyed while still being displayed.
Solution: Add a temporary buffer and destroy it when the next buffer is
attached.
Previously, mpv incorrectly used the %HOME% environment variable on
MinGW to determine the current user’s home directory. This is wrong;
the correct variable to use would be %HOMEPATH%, which would however
still be wrong since application data goes into the application data
directory, not the user’s home. This patch makes it use the local
AppData path instead of reading an environment variable.
This however exposed another problem (which also affected users who
actually had the %HOME% variable set):
b2c2fe7a37 (discussed in issue #95) introduced some changes that
make mpv load user config files from the executable path on Windows.
The problem with this change is that config_dir was still declared
static, so once a config file had been found in the executable path,
it would set config_dir to an empty string, so mpv would dump e.g.
watch_later data straight into the user’s home. This commit also
fixes that.
One side effect of this is that mpv no longer considers the “mpv”
subdirectory in the executable path (that behavior resulted from
the homedir variable always being empty), unless it is somehow
unable to determine the local AppData path.
This is mostly related to the fullscreen behaviour. cecbd8864 introduces an
option to make mpv behave like a OSX user would expect. This commit changes
the Cocoa parts of the code to be consistent with the behaviour on X11. Old
behaviour is still available through the option mentioned in cecbd8864.
There is still custom logic in the cocoa backend and it can probably be moved
to core:
* Don't perform autohide if the mouse is down
* Don't perform autohide outside of the video window
Fixes#218 (by accident)
libquvi 0.4 doesn't allow us listing the formats supported by a
streaming site without doing additional network accesses, so switching
formats was not supported with it. (It's different with libquvi 0.9.)
But the most important case is switching between SD and HD. Usually,
--quvi-format=default will get SD, while --quvi-format=best gives HD.
Use this, and pretend that an URL supported by libquvi 0.4 supports both
of these. "cycle quvi-format" will switch between these. If the user
specifies something else via --quvi-format, this is included in the list
of switchable formats additionally to "default" and "best".
It's annoying for users if you can't get a list of options with --help,
but on the other hand, printing all options would be overkill. So just
mentioned --list-options.
Retrieve per-chapter metadata, but don't do much with it. We just make
the metadata of the _current_ chapter available as chapter-metadata
property. Returning the full chapter list with metadata would be no
problem, except that the property interface isn't really good with
structured data, so it's not available for now.
Not sure if it's worth it, but it was requested via github issue #201.
Consider the cluster used for prerolling contains an insane amount of
subtitle packets. Then the demuxer packet queue would be full of
subtitle packets, and demux.c would refuse to read any further packets -
including video and audio packets, resulting in EOF. Since everything
involving Matroska and subtitles is 100% insane, this can actually
happen.
Fix this by putting a limit on the number of subtitle packets read by
preroll, and throw away any further packets if the limit is exceeded. If
this happens, the preroll mechanism will stop working, but the player's
operation is unaffected otherwise.
Cherry picked from various commits in lua_experiment by ChrisK2.
The metrics of the OSD symbols change slightly, possibly due to the
font editor that was used, and the metrics were not correct to begin
with. (But the real reason seems unknown.) Remove the rescaling of
the OSD font in ASS_USE_OSD_FONT, because the height more or less fits
now. (This change wasn't in the lua_experiment branch.)
run_playloop() is already stuffed enough. This function is still quite
big, but all the other code shares various variables, so it's not as
easy to split.
This option makes the cursor always visible in windowed mode.
Apparently, this is what (some?) Windows and OSX users expect. It's
disabled by default for now.
Restructure the cursor hide logic a bit for this purpose.
Let all key events go through mp_input_feed_key() internally, and also
do double click and MP_INPUT_RELEASE_ALL handling there. Move
check_autorepeat() to where it's actually used.
This caused the OSC to be always visible at startup on X11:
- EnterNotify event send a mouse event to input.c
- OSC has not completely initialized yet, and no mouse area is set
- mouse event is dispatched to "showhide" OSC section
- OSC becomes visible, regardless of mouse position
Fix this by treating the mouse area as empty if it's not set, instead of
infinite as it was before this commit. This means an input section must
set a mouse area to receive mouse events at all. We also have to change
the default section to receive mouse events with the new behavior.
Also, if MOUSE_MOVE is unmapped (or mapped to something that doesn't
parse), and produces no command, the mouse position wouldn't be updated
(because the mouse position is bound to input commands), so we have to
generate a dummy command in this case.
(This matters only for the OSC, On Screen Controller, which isn't merged
yet, so these changes shouldn't have much effect right now.)
Until now, any command was dropped as soon as the input queue was full,
and the command was not an abort command (i.e. a command that exits the
player or goes to the next file).
This could cause issues with key down events (especially mouse buttons)
not being released.
Change it so that key up events can never be dropped. This is a bit
involved, because we know whether a key maps to an abort command only
after interpreting it, and interpreting it changes global state, which
in turn requires undoing the event if the input is dropped. Refactor
the code a bit to move more functionality into interpret_key() to make
this easier.
Remove options which are too obscure and most likely not very useful in
general, or update them to something more modern. Add some comments
about how configuration files work in general.
Even if a subtitle was explicitly loaded with -sub, it was still auto-
loaded (if auto-loading applied to that file). Fix this by explicitly
checking whether a file is already loaded.
The check is maximal naive and just compares the filenames as strings.
The change in find_subfiles.c is so that "-sub something.ass" happens to
work (auto-loading prepended a "./" to it, so the naive filename
comparison check didn't work).
External vobsubs usually come as .idx/.sub pairs. Loading the .idx file
implicitly loads the .sub file, whereas loading the .sub file will kind
of work, but miss important information such as subtitle resolution. Or
in other words, if the .idx file exists, adding the .sub file as track
is useless and confusing.
Explicitly remove .sub file from the auto-load suntitle list in these
cases. Standalone .sub files are still loaded.
We also drop that weird logic that excluded .utf8 files from being
loaded if -subcp was in use. I hope the associated use case didn't make
much sense to begin with. If not, we could still implement it properly,
instead of this weird hack.
Use mp_path_exists() to check for existence of a file (which in turn
uses stat()), instead of opening and closing it. The difference is that
if we don't have sufficient permissions to read the subtitle files, we
will loudly complain. Personally, I prefer this behavior.
Until now, options could be accessed as properties via "options/name",
but the access was read-only. Change it so that write access is possible
in --idle mode. (All options have to support setting options at that
time, simply because of the way MPlayer designed per-file options.
During playback, normal properties take care of changing things,
including things backed by options.)
This is work in progress. There are some issues: at least setting the
"vf" and "af" options won't work for strange reasons.
Pretty useful for people writing userscripts for web browsers. Links starting
with 'mpv://' are forwarded to the mpv OSX bundle. The leading 'mpv://' is
stripped from the recived url and the rest of the string is inserted as is in
the playlist.
The --volume option accepted values up to 10000, but internally, the
value is always clipped to 0-100 range. What makes this even worse is
that --softvol-max suggests that it extends the range of --volume, which
is not the case. (And passing a volume larger than 100 to --volume
didn't even print a warning.)
The really funny thing about this commit is that this code is added on
top of another work around. Basically, subtitle seeking in libavformat
is completely broken. To make it useful, we have to add yet another
workaround.
The basic problem is that libavformat's subtitle seeking code always
uses the stream time base, instead of AV_TIME_BASE if stream index -1 is
passed to the avformat_seek_file() function.
Fixes github issue #216. Hopefully this will be fixed in ffmpeg too at
some point.
This affects MOUSE_MOVE and MOUSE_LEAVE. Both are needed internally
(such as for the OSC), but not really useful for input.conf. Since the
warning has the purpose of notifying the user that a key is unmapped and
what key name to use for setting up a binding in input.conf, the warning
is rather useless in this case. It's also annoying in combination with
the
--no-input-default-bindings option, since that removes the default
bindings to "ignore" for these keys.
When enabling --save-position-on-quit, playback position stored not only
on quit, but in any case playback of a file was stopped. This includes
going to the next file with playlist navigation commands.
After some discussion on IRC, it turned out that nobody thought this was
good behavior. Disable it, and really make it save only on quit.
Maybe the option is useless now, as the user could remap the CLOSE_WIN
key binding. On the other hand, CLOSE_WIN sounds and _is_ a bit obscure.
Before this commit, the player didn't write resume info if the playback
position was within the first or last percent of the file.
This was sometimes annoying, and with playlist resume can lead to
unintuitive behavior. (It wouldn't resume the playlist if the currently
played file was at 0-1% or 99-100%, even if you were in the middle of a
playlist.)
Moreover, the "percent > 99" check is a bit bogus anyway, because 100
(in integer) is rarely reached.
Drop the check, and make sure using --save-position-on-quit won't write
resume info when reaching EOF. The latter check is needed to make sure
playback of the file starts at beginning when playing it again after
EOF.
The previous check just searched for a "://" substring. This was quite
bad, because "://" can be a valid part of a path. Later, I added
special handling for filenames starting with "." and "/", so that you
could reliably pass arbitrary filenames to mpv, no matter how messed
up they were.
Even though it doesn't really matter in practise anymore, this is still
crap, so add a more reliable check instead.
This includes the case of passing multiple files to command line
(internally this is the same as loading a playlist).
Resuming works by finding the first playlist entry that can be resumed.
Alternative implementations would be possible, such as hashing the
playlist contents. But this implementation is simpler, and doesn't have
the disadvantage that changes to the playlist (like appending entries)
will throw away the resume point.
This makes loading large playlists a bit slower, because it has to look
into ~/.mpv/watch_later/ for every entry. Loading a 15000 entries
playlist now increases from 150ms to 400ms. Considering you rarely load
playlists this big with mpv (because it's impractical considering the
terminal and non-GUI nature of the player), this is probably ok.
Undo URL percent encoding if the filename appears to be an URL. This
will fix display of the actual filename in some cases.
We don't put any effort into checking whether the URL is really percent
encoded, because we don't really know how the protocol handler is going
to interpret the URL. For stream_lavf, we probably can't know. Still,
from the perspective of this commit, it seems to make sense to assume
they are percent encoded.
icon_size is the number of array items of type long, not bytes. Change
the type of icon_size to int, because size_t makes you think of byte
quantities too quickly.
As an unrelated change, change the (char *) cast to (unsigned char *),
because it matches the common XChangeProperty idiom better.
MPlayer handles this correctly, because MPlayer still has the FourCC
codec dispatch (codecs.conf). We need to handle this case specially,
because the libavformat rawvideo decoder will of course not eat mjpeg.
mjpeg is the only supported format, though. (Even MPlayer needs to
convert between V4L2 formats and MPlayer FourCCs, and mjpeg is the only
non-raw format.)
The option list contains an empty string member with this option value,
so ignore that. I'm not sure whether the option list should maybe be
empty in this case, but it could be the wrong thing in case of other
options.
This happens by default with pausing: if a file was paused when doing
quit_watch_later, then resume and unpause, then the file played after
that would start in paused mode. This is because the pause option is
backed up at thr wrong place, so it backs up the state from resuming,
instead of whatever it was set to before that.
This prevents keys to become stuck due to changing keyboard modifiers during
the key down. Not the prettiest approach but event `x11_common` does it like
this.
Fixes#210
The png file added to etc/ are taken from the link mentioned in commit
303096b, except that they have been converted to 16 bit, sRGB (with
color profile info dropped, if there was one), and transparent pixels
reset for better compression.
The file x11_icon.bin is generated by gen-x11-icon.sh. I'm adding it to
the git repo directly, because the script requires ImageMagick, and we
don't want to make building even more complicated.
The way how this is done is basically a compromise between effort
required in x11_common.c and in gen-x11-icon.sh. Ideally, x11_icon.bin
would be directly in the format as required by _NET_WM_ICON, but trying
to write the binary width/height values from shell would probably be a
nightmare, so here we go.
The zlib code in x11_common.c is lifted from demux_mkv.c, with some
modifications (like accepting a gzip header, because I don't know how to
make gzip write raw compressed data).
This allows other threads to use mp_input_put_key without blocking if the
playloop is doing the 500ms select call (i.e.: during pause).
Makes Cocoa GUI responsive again (regression since 2d363c3).
If the input section is enabled with MP_INPUT_ALLOW_VO_DRAGGING, then
the VO will be allowed to drag the window with mouse button down +
mouse move even if the mouse is inside the section's mouse area.
If the mpv window is unfocus, clicking on the OSC should focus the
window (done by the window manager) and allow interaction with the OSC.
But somehow X sends a spurious LeaveNotify event, immediately followed
by an EnterNotify event. This happens at least with IceWM. The result is
that the OSC will disappear (due to receiving MOUSE_LEAVE). The OSC will
stay invisible, because EnterNotify isn't handled, and there's nothing
that could make the OSC appear again.
Solve this by handling EnterNotify. We cause a redundant MOUSE_MOVE
event to be sent, which triggers the code to make the OSC visible. We
have to remove the code from input.c, which ignores redundant mouse move
events.
Since the code ignoring redundant mouse move events is still needed on
Windows, move that code to w32_common.c. The need for this is documented
in the code, also see commit 03fd2fe. (The original idea was to save
some code by having this code in the core, but now it turns out that
this didn't quite work out.)
Normally, moving the mouse outside of the mouse area of an input section
will send mouse events somewhere else (because input section mouse areas
are similar to windows/widgets in real GUI toolkits). This was done even
if a mouse button was held down. This is quite different from how GUI
toolkits behave.
Change the code so that if a mouse button is down, the mouse area of the
current input section can't be left. Releasing the mouse button (while
the mouse pointer is outside of the mouse area) will actually leave the
mouse area.
As a side-effect, this commit also tests more often whether the current
mouse input section is valid. This is needed to make releasing a mouse
button trigger the mouse input section change.
These keys can be found on various "multimedia" and "internet" keyboard.
X defines many keycodes, so I'm not adding all, just what I found on my
own keyboard.
Other key codes can be added on request.
This is basically a libavcodec API oddity: it can happen that
avcodec_decode_audio4() returns 0 (meaning 0 bytes were consumed). It
requires you to feed the complete packet again to decode the full
packet, and to successfully decode the following packets.
We ignored this case with the argument that there's the danger of an
endless decode loop (because nothing of that packet is apparently
decoded, so it would retry forever), but change it in order to decode
mpc8 files correctly.
Also add some comments to explain the mess.
I suspect most users will just copy etc/input.conf when they want to
remap some default bindings. But usually this means the user even copies
bindings he doesn't care about, and it's better if the user maps only
the bindings in his input.conf the user intends to remap.
Comment all bindings in etc/input.conf. Since this file also defines the
builtin defaults and is baked into the mpv binary, we have to do
something to get them anyway, even though they are commented. Do this by
having input.c "uncomment" the bindings in the baked in input.conf. (Of
course this is done only for the builtin config, not configs loaded from
disk.)
The previous code was locking only the input queue. That was too weak since it
didn't protect the input_ctx data structure. So remove the locking on the queue
and lock all the public functions that interact with the input_ctx.
The private functions and public functions that do not act on the input_ctx
(there are quite some functions doing mp_cmd manipulations for instance) are
not locked.
Some changes by wm4. Use a recursive mutex, and restructure some code to be
less annoying with locks, such as converting code to single return, or
making use of the recursive mutex.
Generate a mouse down event on the first click so that one can interact
with the OSC directly as opposed to wasting the first click in order to focus
the window.
I would like to thank Chris Ward (@tenzerothree, http://tenzerothree.com/) for
working on the art for these icons and bringing some eye candy to the project.
The PSDs made by Chris are available on our Dropbox [1], along with the exports
I made to create OSX and Windows icons. The PSDs are almost completly vector
and all the resolutions look really similar, except the 16px favicon which was
handcrafted to look better and more recognizeable on the smaller pixel budget.
For Mac OS X the icons were created using iconutils on the PNGs iconsets
exported from the PSDs. These even support retina resolutions (except 512@2x).
For Windows the .ico file was created with imagemagick. The included images
are 16px, 24px, 32px, 48px 64px, 256px. These are the resolutions listed on
MSDN for supporting Windows XP [2] and Windows versions based on Aero [3].
Only 32bit PNGs were used since it is 2013.
For Linux nothing changed yet, even though @wm4 talked about using the PNGs
directly there. This will probably be dealt with in a later commit.
[1]: https://www.dropbox.com/sh/yelfoj9tbft7o06/A8vOT6JKaG
[2]: http://msdn.microsoft.com/en-us/library/ms997636.aspx
[3]: http://msdn.microsoft.com/en-us/library/aa511280.aspx
This probably has been broken since bbc865a: a test was added that uses
a FBO, but it's always run, even if FBOs were not detected. On the other
hand, fbotex_init() just runs into an assert. Fix the test that
triggered this condition, and make fbotex_init() "nicer" by just failing
if FBOs are not available.
Port it from playlist_parser.c to demux_playlist.c. Also, change the m3u
parser to drop whitespace from the trailing part of the line (will make
it work properly with windows line endings).
(I hoped that this would make MMS URIs with http instead of mmsh
prefixes work, but it doesn't. Instead, it leads to a playlist loop. So
solving this issue would require a change in ffmpeg, probably.)
Until now, stream_peek() read only the bare minimum it had to read from
the stream. But this could cause problems, such as being very
inefficient when peeking a lot, or conflicting with ability to seek
back. (The latter issue can be caused by peeking a few bytes, and then
doing a stream_read() with a size that is 1 byte longer: this would read
the peeked data, then call stream_fill_buffer(), which throws away the
previously peeked bytes - so you can't seek back anymore. This is
mitigated by a hack in demux_open(): it peeks a full buffer, to avoid
that peeking/reading during demuxer probing [or before that, in a stream
filter] can cause the buffer to be dropped.)
The original condition was too weak, requiring only the header. The header is
installed is FFmpeg regardless of the presence of VDA on the system, so just
perform a check on the `ff_vda_create_decoder` function.
Make use of opaque regions on non-alpha formats. This allows the compositor to
improve the drawing of the surface, because he can discard everything behind
the window when drawing.
Odd video sizes if pixel formats with chroma subsampling and PBOs were
used, garbage was rendered. This was because the PBO path created
buffers with an unpadded size, and then tried to upload a padded
image to it. Fix it by explicitly setting the padded size. (As with
the non-PBO path, we rely that image allocations are somehow padded,
which is normally the case.)
Apparently, it is popular to store large files in uncompressed rar
archives. Extracting files is not practical, and some media players
suport playing directly from uncompressed rar (at least VLC and some
DirectShow components).
Storing or accessing files this way is completely idiotic, but it is
a common practice, and the ones subjected to this practice can't do
much to change this (at least that's what I assume/hope). Also, it's
a feature request, so we say yes.
This code is mostly taken from VLC (commit f6e7240 from their git tree).
We also copy the way this is done: opening a rar file by itself yields
a playlist, which contains URLs to the actual entries in the rar file.
Compressed entries are simply skipped.
Add a stream filter concept, in which streams can be opened on top of
an underlying "source" stream. Change the open code to make this
easier, and also to account for some mechanisms that will be needed
for this.
The following commit will add stream_rar, which contains such a stream
filter.
This is really not needed. While we really can't take a loaded buffer
over to the cache, there's no reason why the cache couldn't read this
buffer normally.
On the other hand, this code could cause trouble when probing from a
stream before the cache has been enabled.
The way the url_options field was handled was not entirely sane: it's
actually a flexible array member, so it points to garbage for streams
which do not initialize this member (it just points to the data right
after the struct, which is garbage in theory and practice). This was
not actually a problem, since the field is only used if priv_size is
set (due to how this stuff is used). But it doesn't allow setting
priv_size only, which might be useful in some cases.
Also, make the protocols array not a fixed size array. Most stream
implementations have only 1 protocol prefix, but stream_lavf.c has
over 10 (whitelists ffmpeg protocols). The high size of the fixed
size protocol array wastes space, and it is _still_ annoying to
add new prefixes to stream_lavf (have to bump the maximum length),
so make it arbitrary length.
The two changes (plus some more cosmetic changes) arte conflated into
one, because it was annoying going over all the stream implementations.
Modeled after the old playlist_parser.c, but actually new code, and it
works a bit differently.
Demuxers (and sometimes streams) are the component that should be used
to open files and to determine the file format. This was already done
for subtitles, but playlists still use a separate code path.
af_str2fmt_short(), which is used by the command line option parser,
allowed passing a hex number. The user could set arbitrary integers as
internal audio formats, even formats which don't exist or make no sense.
This is not very useful, so get rid of it.
Having to use -1 for that is generally quite annoying.
Audio formats are created from bitmasks, and it can't be excluded that
0 is not a valid format. Fix this by adjusting AF_FORMAT_I so that it
is never 0. Along with AF_FORMAT_F and the special formats, all valid
formats are covered and guaranteed to be non-0.
It's possible that this commit will cause some regressions, as the
check for invalid audio formats changes a bit.
A wayland output based on shared memory. This video output is useful for x11
free systems, because the current libGL in mesa provides GLX symbols. It is also
useful for embedded systems where the wayland backend for EGL is not
implemented like the raspberry pi.
At the moment only rgb formats are supported, because there is still no
compositor which supports planar formats like yuv420p. The most used compositor
at the moment, weston, supports only BGR0, BGRA and BGR16 (565).
The BGR16 format is the fastest to convert and render without any noticeable
differences to the BGR32 formats. For this reason the current (very basic)
auto-detection code will prefer the BGR16 format. Also the weston source code
indicates that the preferred format is BGR16 (RGB565).
There are 2 options:
* default-format (yes|no) Which uses the BGR32 format
* alpha (yes|no) For outputting images and videos with transparencies
The obtained information from the shm listener isn't used by anything and is
also wrong now in wayland git master branch because of new shm formats which
need a different way of saving the supported formats.
Moves a good chunk of the resizing code to wayland_common.c. This makes it
possible to share it with future video drivers.
It doesn't resizit it immediatly, it calcutlates the new position and size and
then shedules a resizing event. This removes the ugly callback and void pointer
from the wayland data structure.
Regression since 18b6c01d92. That commit changed the colorspace handling to
always reinit the video output. Since the CVPixelBuffers are lazily created,
VOCTRL_SET_YUV_COLORSPACE was always called when the CVPixelBufferRef was NULL.
Since CoreVideo functions do not complain when called on NULL, no one noticed
that CVBufferSetAttachment, which stored the color matrix meta data was called
on NULL.
Until now, video output levels (obscure feature, like using TV screens
that require RGB output in limited range, similar to YUY) still required
handling of VOCTRL_SET_YUV_COLORSPACE. Simplify this, and use the new
mp_image_params code. This gets rid of some code. VOCTRL_SET_YUV_COLORSPACE
is not needed at all anymore in VOs that use the reconfig callback. The
result of VOCTRL_GET_YUV_COLORSPACE is now used only used for the
colormatrix related properties (basically, for display on OSD). For
other VOs, VOCTRL_SET_YUV_COLORSPACE will be sent only once after config
instead of twice.
This affects VOs which just reuse the mp_image from draw_image() to
return screenshots. The aspect of these images is never different
from the aspect the screenshots should be, so there's no reason to
adjust the aspect in these cases.
Other VOs still need it in order to restore the original image
attributes.
This requires some changes to the video filter code to make sure that
the aspect in the passed mp_images is consistent.
The changes in mplayer.c and vd_lavc.c are (probably) not strictly
needed for this commit, but contribute to consistency.
mp_image_set_params() doesn't check whether the colorspace parameters
are consistent (e.g. setting YUV colorspaces with RGB formats), and
shouldn't need to.
The way this was added to FFmpeg is less than ideal, because it requires
text parsing in the Matroska demuxer. But in order to use the FFmpeg
webvtt-to-ass converter, we still have to mimic this in some way. We do
this by putting the parsing into sd_lavc_conv.c, before the subtitle
packet is passed to libavcodec. At least this keeps the ugliness out of
unrelated code.
There is some change that FFmpeg will fix their design eventually.
Instead of rewriting the parsing code, we simply borrow it from FFmpeg's
Matroska demuxer.
cocoa_common contains some locking calls to support video outputs that support
live resizing (at this moment only vo=opengl).
These should not be used unless the VO declares it is multithreaded by
registering the resize_redraw callback used for live resizing.
Fixes#200
Use the new MP_ macros for some AOs instead of mp_msg.
Not all AOs are converted, and some only partially. In some cases, some
additional cosmetic changes are made.
Otherwise, this would just try to demux a good chunk of the file, even
though the operation can't succeed anyway.
This caused some pretty strange issues, where perfectly valid use cases
would print a "Too many packets in the demuxer packet queue..." message.
The rawaudio demuxer read one frame per packet, basically a few bytes,
which caused insane overhead. (I found this when I couldn't play raw
audio without dropouts when using -v, which printed a line per packet
read.)
Fix this and read 1 second of audio per packet. This is a regression
since cfa5712 (merging of demux_rawaudio and demux_rawvideo).
Instead of always skipping in STREAM_BUFFER_SIZE blocks, allow an
arbitrary size. This allows - in theory - faster forward seeking in
pipes.
(Maybe not a very significant change, but it reduces the number of
things that depend on STREAM_BUFFER_SIZE for no good reason. Though
we still use that value as minimum read size.)
stream_file.c contains some code meant for forward seeking with pipes.
This simply reads data until the seek position is reached. Move this
code to stream.c. This stops stream_file from doing strange things
(messing with stream internals), and removes the code duplication too.
We also make stream_seek_long() use the new skip code. This is shorter
and much easier to follow than the old code, which basically did strange
things.
Decoding H264 using Video Decode Acceleration used the custom 'vda_h264_dec'
decoder in FFmpeg.
The Good: This new implementation has some advantages over the previous one:
- It works with Libav: vda_h264_dec never got into Libav since they prefer
client applications to use the hwaccel API.
- It is way more efficient: in my tests this implementation yields a
reduction of CPU usage of roughly ~50% compared to using `vda_h264_dec` and
~65-75% compared to h264 software decoding. This is mainly because
`vo_corevideo` was adapted to perform direct rendering of the
`CVPixelBufferRefs` created by the Video Decode Acceleration API Framework.
The Bad:
- `vo_corevideo` is required to use VDA decoding acceleration.
- only works with versions of ffmpeg/libav new enough (needs reference
refcounting). That is FFmpeg 2.0+ and Libav's git master currently.
The Ugly: VDA was hardcoded to use UYVY (2vuy) for the uploaded video texture.
One one end this makes the code simple since Apple's OpenGL implementation
actually supports this out of the box. It would be nice to support other
output image formats and choose the best format depending on the input, or at
least making it configurable. My tests indicate that CPU usage actually
increases with a 420p IMGFMT output which is not what I would have expected.
NOTE: There is a small memory leak with old versions of FFmpeg and with Libav
since the CVPixelBufferRef is not automatically released when the AVFrame is
deallocated. This can cause leaks inside libavcodec for decoded frames that
are discarded before mpv wraps them inside a refcounted mp_image (this only
happens on seeks).
For frames that enter mpv's refcounting facilities, this is not a problem
since we rewrap the CVPixelBufferRef in our mp_image that properly forwards
CVPixelBufferRetain/CvPixelBufferRelease calls to the underying
CVPixelBufferRef.
So, for FFmpeg use something more recent than `b3d63995` for Libav the patch
was posted to the dev ML in July and in review since, apparently, the proposed
fix is rather hacky.
Regression since ff3b98d11c. The window positioning code relied on the
visibleFrame's height without taking into account the dock's presence.
Also moved the constraining code to the proper method that overrides the
original NSWindow behaviour. This avoids having to check for border since the
constraining is performed by Cocoa only for titled windows.
Fixes#190
Too many people thought "D:" really meant number of dropped frames. But
it's actually the number of frames where the playloop thought it'd be
a good idea to drop them. Of course this does nothing if frame dropping
is disabled, but even with normal frame dropping, this doesn't indicate
whether a frame was _really_ dropped. (Looks like libavcodec doesn't
even give us this information reliably? The decode function can return
no frame in case of codec delay due to threading and such.)
Originally, the objective of this commit was changing --edition to be
1-based, but this was cancelled. I'm still leaving the change to
demux_mkv.c though, which is now only of cosmetic nature.
We don't need to store the offsets of the options corresponding to the
properties, because the option-property bridge knows about this already.
The check against 1000 was for the case if e.g. the --brightness option
is not used. Always overwrite the option value instead, both when
querying and setting the property. (This is needed to make the settings
persistent even if vf_eq is used and the video chain is reinitialized.)
This commit assumes that VFCTRL_SET_EQUALIZER is always paired with
VFCTRL_GET_EQUALIZER (likewise for VOCTRL), which is the case.
Using -vf eq and changing brightness, contrast, etc. using key bindings
with e.g. "add brightness 1" didn't work well: with step width 1, the
property gets easily "stuck". This is a rounding problem: e.g. setting
gamma to 3 would actually make it report that gamma is set to 2, so
the "add" command will obviously never reach 3 with a step width of 1.
Fix this by storing the parameters as integers.
This was broken in cac7702. This commit effectively changed these
properties to use the value as reported by vf_eq, instead of the
previously set value for the "add" command. This was more robust,
but not very correct either, so we keep the new behavior and make
vf_eq report its parameters more accurately.
The display, window, keyboard and cursor structures are now cleanly and
logically separated. Also could prevent a future bug where no shm format is set
when the cursor image is loaded (Never happened until now).
Add --video-align-x/y, --video-pan-x/y, --video-scale options and
properties. See the additions to the manpage for description and
semantics.
These transformations are intentionally done on top of panscan. Unlike
the (now removed) --panscanrange option, this doesn't affect the default
panscan behavior. (Although panscan itself becomes kind of useless if
the new options are used.)
This option allowed you to extend the range of the panscan controls, so
that you could essentially use it to scale the video. This will be
replaced by a separate option to set the zoom factor directly.
See github issue #194.
Unfortunately, this breaks the property that going back in the playlist
always works as expected. This changes, because the playlist_prev
command will work on the reshuffled playlist, instead of loading the
previously played files in order. If this ever becomes an issue, I
might revert this commit.
Now the code does the same as the original MPlayer VAAPI patch, instead
of trying to map the profiles exactly.
See previous commit for justification and discussion.
Instead, do what MPlayer did all these years. It worked for them, so
there's probably no reason to change this.
Apparently fixes playback with some files, where the VDPAU decoder does
not formally support a profile, but decoding works with a more powerful
profile anyway.
Though note that MPlayer did this because it couldn't do it in a better
way (no decoder reported profiles available when creating the VDPAU
decoder), so it's not entirely clear whether this is a good idea. An
alterbative implementation might try to map the profiles exactly, and
do some fall backs if the exact profile is not available. But this
hack-solution works too.
Apparently this was dropped some years ago, but judging from MPlayer's
handling of this, the original code wasn't so great anyway. The new
code handling clearing of panscan borders correctly, and integrates
better with the YUV path. (Although the VDPAU API sure makes this
annoying with its separate surface types for RGB.)
Note that we create 5 surfaces for some reason - I don't think this
makes too much sense (because we can't use the deinterlacer with RGB
surfaces), but at least it reduces the amount of differences with
the YUV code path.
Clearing the borders is done by drawing a single black pixel over the
window. This sounds pretty dumb, but it appears to work well, and
there is no other API for that. (One could try to use the video mixer
for this purpose, since it has all kinds of features, including
compositing multiple RGBA surfaces and clearing the window background.
But it would require an invisible dummy video surface to make the
video mixer happy, and that's getting too messy.)
When panscan was used, i.e. the video is cropped to make the video fill
the screen if video and screen aspects don't match, screenshots
contained only the visible region of the source video, stretched to
original video size.
Consider:
mpv file1.mkv file2.mkv
and file1.mkv is restored from an earlier session when quit_watch_later
was used. Then all restored options were reset when file2.mkv is played,
even if the user changed them during playback. This affects for example
the fullscreen setting.
Make it so that after finishing a resumed file, the previously restored
settings are not reset again. (Which means only resuming will forcefully
overwrite the settings.)
If close to chapter start, skipping back goes to previous chapter (no change).
If more than <threshold> seconds in, skipping back will now go to the beginning
of the current chapter instead.
The threshold is set by the new option --chapter-seek-threshold and defaults to
5 seconds. A negative value disables the new functionality.
The VDPAU default colorkey, although it seems to be driver specific, is
usually green. This is a pretty annoying color, and you usually see it
briefly (as flashes) if the VDPAU window resizes.
Change it to some shade of black. The new default color is close to what
MPlayer picks as colorkey (and apparently it worked well for them):
VdpColor vdp_bg = {0.01, 0.02, 0.03, 0};
Since our OPT_COLOR can set 8 bit colors only, we use '#020507' instead,
which should be the same assuming 8 bit colors.
Obviously, you can't use black, because black is a way too common color,
and would make it too easy to observe the colorkey effect when e.g.
moving a terminal with black background over the video window.
Formally, this sets the "background color" of the presentation queue.
But in practice, this color is also used as colorkey.
This commit doesn't change the VDPAU default yet.
Not actually useful. This would break whenever a new text subtitle
format would be added, which requires a binary->text transformation.
(mov_text is one such format; disable it.) In general, we would have
to know which packet formats are binary, which we don't, so the only
reasonable way to handle this is a white list.
One example in ao.rst used the old syntax with mangled device names.
Fix it. Mention some shell related caveats. Explicitly mention the
change of device name syntax in changes.rst, because it seems to be
a common issue.
Normally, --subcp always forces conversion. This really always forces
conversion, even if the UTF-8 check on the input succeeds.
Extend the --subcp to allow codepages as fallback if UTF-8 doesn't
work. So, for example --subcp=utf8:cp1250 will use UTF-8 if the input
looks like UTF-8, and will fall back to use cp1250 if the UTF-8 check
fails.
I think this should actually be the default, but on the other hand,
this changes the semantics of the option, and a user would actually
expect --subcp to force conversion, rather than silently using UTF-8
if that happens to work.
Don't accept overlong sequences. Don't accept codepoints past the
maximum unicode codepoint. Don't accept the UTF-16 surrogate codepoints.
I'm not sure if there are more codepoints that are defined to be
invalid, but we just want to make libavcodec happy, so this is enough.
(libavcodec's subtitle converter checks for valid UTF-8 and throws up
and dies if it's not - now we want to use bstr_sanitize_utf8_latin1() to
force valid UTF-8, so the strictness of our UTF-8 parser has to match at
least that of the libavcodec's check.)
I'm not sure whether the min test is actually 100% correct.
Note that libavcodec also treats BOM codepoints as invalid. This is
definitely a bug: the BOM is really just "zero-width non-breaking space"
redefined by Microsoft, but it is perfectly valid to appear in the
middle of a string. Official Unicode has merely deprecated the old
usage of the BOM codepoint, and didn't make it illegal. Besides, the
string could be from the start of a file, so even this check doesn't
make sense even with libavcodec's insane logic. We don't copy this bug.
Broken UTF-8 in this context means we treat it as UTF-8, but we also
interpret broken UTF-8 sequences as Latin1.
Also, run our own UTF-8 check function before the charset detectors.
This prevents from ENCA's UTF-8 check possibly messing up (like
detecting 7-bit clean UTF-8 as ASCII, or other things). It also takes
care of UTF-8 detection if no charset detector (ENCA, libguess) is
compiled in, and it lets us deal better with cut-off UTF-8 sequences.
Libav's <libavcodec/vdpau.h> header uses some libavocdec symbols without
forward-declaring them and without including the headers declaring them.
FFmpeg's header for this is fine.
About this issue, it would be better if the surfaces could be allocated
with the real size, and the vdpau video mixer could be created with that
size as well. That would be a bit hard, because the real surface size
had to be communicated to vdpau. So I'm going with this solution. vaapi
seems to be fine with either surface size, so there's hopefully no
problem.
Instead of passing AVFrame. This also moves the mysterious logic about
the size of the allocated image to common code, instead of duplicating
it everywhere.
The fix_overlaps_and_gaps() function in dec_sub.c fixes small gaps or
overlaps between subtitle events. However, sometimes it could happen
that the corrected subtitle events could overlap by 1ms due to bad
rounding, making libass shift subtitles to reduce collisions. (The
second subtitle will be shown above the previous one, even if both
subtitles are visible only for 1ms.)
sd_ass.c rounds the timestamps when converting to integers for unknown
reasons. I think it would work fine without that rounding, but since
I have no clue why it rounds, and since it could be needed to ensure
correct timestamps with ASS subtitles demuxed from Matroska, I'd rather
not touch it. So the solution is to use already rounded timestamps to
calculate the new subtitle duration in fix_overlaps_and_gaps().
See github issue #182.
This adds precise scrolling support. I ran some tests and it seems a little
bit smoother and well.. precise. The defaults are rebindable using: AXIS_UP,
AXIS_DOWN, AXIS_LEFT and AXIS_RIGHT.
If pthreads are enabled the input queue accesses are regulated by acquiring
a mutex. This is useful for platforms like OS X, where the events are created
in the cocoa thread and added to the queue to then be dequeued in the playloop
thread.
This reverts commit 689a25003f, with some
adjustments to code that was added after that commit.
I just messed up big time. We don't need this, and in fact the commit
confused straight and premultiplied alpha at one point (just a simple
inverted condition due to an oversight), which is why it looked like
it was working.
In commit 2827295 I wrote:
Also, libva can't decide whether it accepts straight or premultiplied
alpha for OSD sub-pictures [...]
That was just me messing up and being severely confused by my own bugs.
VA API uses premultiplied alpha, which by the way is nice and
thoughtful of the VA API devs.
Well, this was stupid. But in the end, I'm glad that I could actually
reduce codesize by a good amount again.
This is based on the MPlayer VA API patches. To be exact it's based on
a very stripped down version of commit f1ad459a263f8537f6c from
git://gitorious.org/vaapi/mplayer.git.
This doesn't contain useless things like benchmarking hacks and the
demo code for GLX interop. Also, unlike in the original patch, decoding
and video output are split into separate source files (the separation
between decoding and display also makes pixel format hacks unnecessary).
On the other hand, some features not present in the original patch were
added, like screenshot support.
VA API is rather bad for actual video output. Dealing with older libva
versions or the completely broken vdpau backend doesn't help. OSD is
low quality and should be rather slow. In some cases, only either OSD
or subtitles can be shown at the same time (because OSD is drawn first,
OSD is prefered).
Also, libva can't decide whether it accepts straight or premultiplied
alpha for OSD sub-pictures: the vdpau backend seems to assume
premultiplied, while a native vaapi driver uses straight. So I picked
straight alpha. It doesn't matter much, because the blending code for
straight alpha I added to img_convert.c is probably buggy, and ASS
subtitles might be blended incorrectly.
Really good video output with VA API would probably use OpenGL and the
GL interop features, but at this point you might just use vo_opengl.
(Patches for making HW decoding with vo_opengl have a chance of being
accepted.)
Despite these issues, decoding seems to work ok. I still got tearing
on the Intel system I tested (Intel(R) Core(TM) i3-2350M). It was also
tested with the vdpau vaapi wrapper on a nvidia system; however this
was rather broken. (Fortunately, there is no reason to use mpv's VAAPI
support over native VDPAU.)
This is for VAAPI support. VAAPI does not support premultiplied alpha
for OSD. (Normally, we prefer premultiplied, because it has better
behavior on scaling.)
I'm not sure whether blending in the ASS->RGBA part is correct and I
didn't test it extensively.
Change how the HW decoding stuff is organized, the way it's initialized
in particular. Instead of duplicating the list of supported codecs for
hwaccel decoders, add a probe function which allows each decoder to
report whether it supports a given codec.
Add an "auto" choice to the --hwdec option, which automatically enables
hardware decoding if libavcodec and/or the VO supports it.
What mpv prints on the terminal changes a bit. Now it will just print
a single line whether hw decoding is used or not (and nothing at all if
no hw decoding at all was requested). The pretty violent fallback from
hw decoding to software decoding is still quite verbose and evil-looking
though.
2013-08-11 23:59:18 +02:00
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Alessandro Ghedini