It sometimes happens on exit, and it's probably a bad idea. If the
process hangs on exit (possibly due to stupid hardcoded timeouts it's
doing), mpv will also hang now, unfortunately.
(cherry picked from commit be9bf4cc7c)
It appears some WMs have a problem with out method of setting initial
fullscreen mode. We assume that if the window's _NET_WM_STATE includes
_NET_WM_STATE_FULLSCREEN before mapping the window, the WM will show it
as fullscreen at mapped. EWMH doesn't say anything that this should
work, although one could argue that it's implied.
In any case, since it's not standard behavior without at least some
doubt, it's probably a good idea to try the "old" method as well.
Fortunately, it should be idempotent.
See #1937, #1920.
(cherry picked from commit a4f3df5970)
Arch linux is about to update to lua 5.3.x, but lua 5.2.x will be
provided by package lua52, which contains pkg-config file lua52.pc.
Signed-off-by: wm4 <wm4@nowhere>
(cherry picked from commit 5b085fd8b1)
Some code always calls vo_event(), even with event==0, which leads to
immediate wakeup, which in turn causes the function to be called again.
This would burn CPU, which was especially noticeable when paused.
(cherry picked from commit 0bdef9979f)
Gives the VOs more flexibility.
gl_video.c already ignores the timing info if no interpolation is
active, so this requires no further changes.
(cherry picked from commit af157db7e5)
There is not much of a reason to have these wrappers around. Use POSIX
standard functions directly, and use a separate utility function to take
care of the timespec calculations. (Course POSIX for using this weird
format for time values.)
(cherry picked from commit 92b9d75d72)
The main reason for this was compatibility; but some associated problems
have been solved in the previous commit.
(cherry picked from commit ca9964a4fb)
As usual, we prefer plain C11 names and semantics, and have to emulate
them if C11 atomics are not available.
For the non-atomic fallback (which is just there to make code compile in
situations the atomic property is not overly important), we require a
gross hack to make the generic macros work without using compiler-
specific extensions.
(cherry picked from commit cc24ec5b3c)
The first one (for VOCTRL_GET_DISPLAY_FPS) could have led to undefined
behavior if the FPS was unknown. The second is for general symmetry.
(cherry picked from commit 10149f68a5)
Now among other things panscan can be changed during playback.
Unfortunately, it flickers. The issue is that reconfig() clears the
framebuffer. Removing the clearing shows that the "unused" parts of
the picture are not cleared - even though OSD could render there. As
such, this is a separate issue.
(cherry picked from commit 5631060569)
When running with --panscan=1, this could crash - because the current
frame was reduced in size each time the image was redrawn, which would
result in a failed assertion the second time it's drawn.
(cherry picked from commit 859ddc9906)
Drop mp_chmap_diff() (which is unused too now), and implement
mp_chmap_diffn() in a slightly simpler way. (Too bad there is no
standard function for counting set bits.)
(cherry picked from commit 00130651da)
Instead of somehow having 4 different cases with each their own weight,
do it with a single function that decides which channel layout is the
better fallback.
This is simpler, and also introduces new (fixed) semantics. The new test
added to test/chmap_sel.c actually works now. This is a mixed case with
no perfect upmix or downmix, but the better choice is the one which
loses the least channels from the original layout.
One test also changes. If the input is 7.1(wide-side), and the available
layouts are 7.1 and 5.1(side), the latter is now chosen instead of the
former. This makes sense: both layouts contain 6 out of 8 channels from
the original layout, but the 5.1(side) one is smaller. This follows the
general logic. The 7.1 layout has FLC/RLC speakers instead of BL/BR,
and judging by the names, "front left center" is completely different
from "back left". If these should be exchangeable, a separate exception
would have to be added.
(cherry picked from commit 3560a50029)
Reuse MP_SPEAKER_ID_NA for this. If all mp_chmap entries are set to NA,
the channel layout has special "unknown channel layout" semantics, which
are used to deal with some corner cases.
(cherry picked from commit 55e777f10b)
Remove the requirement from mp_chmap that speaker entries must be
unique. Use this to get rid of all the redundant NA speaker IDs.
(cherry picked from commit b91b4944bd)
While mpv has no internal equivalent representation, they can still be
used as physical CoreAudio formats. Thus this label is confusing.
(cherry picked from commit 1bcb82ec93)
Sometimes, ALSA will return channel layouts with padded channels (NA
speakers). Use them instead of failing.
This still includes the old "braindeath" code to retry with a layout
without NA channels. This might be helpful for performance, and also the
padded channel layout string looks confusing.
To be fair, I have not encountered a case yet which would really need
this, and for which the old "braindeath" code did not fix it.
(cherry picked from commit 85fc6b2a05)
One side effect is that the warning about too many channels goes away,
and is replaced with printing the ALSA channel map as "unknown".
(cherry picked from commit d577872a28)
volatile barely means anything.
The polling is kind of bad too, but relatively harmless as device
opening/closing is a rare event, and the format change is not expected
to take long.
Remove the pointless talloc call too (must have been a leftover
from previous refactoring).
(cherry picked from commit 4444ff48fa)
No reason to keep them separate. It's an artifact from the old
ao_coreaudio.c, which kept usage of two different APIs in the same file.
Removes a forward reference too.
(cherry picked from commit 32bc61ae07)
This should fix some crashes due to dangling pointers.
The problem was that with_cocoa_lock_on_main_thread() is asynchronous.
It will not wait until it is finished. In the uninit case, this means
the VO could be deallocated and destroyed while cocoa was still running
uninit code.
So simply wait until it is done by using dispatch_sync(). There were
concerns that this could introduce a deadlock by the main thread trying
to wait for something on the VO thread. But from what I can see, this
never happens, and even if it does, it would crash anyway since the VO
is already gone.
One remaining worry is the video_resize_redraw_callback. From what I can
see, it still can mess things up, and will need a more elaborate fix.
(cherry picked from commit e777756301)
Instead of trying to use af_format_conversion_score() (which tries to be
all kinds of clever), just compare the raw bits as a quality measure. Do
this because otherwise, weird formats like padded 24 bit formats will be
excluded, even though they might be the highest precision formats for
some hardware.
This means that for now, the user would have to check whether the format
is usable at all before calling ca_asbd_is_better(). But since this is
currently only used for ao_coreaudio.c and for the physical format, it
doesn't matter.
If coreaudio-exclusive should get PCM support, the best would be to
revert this change, and to add support for 24 bit formats directly.
(cherry picked from commit 4ffcf2531b)
Some time ago, a mechanism was added for automatically removing PCM-only
filters if the input format is spdif.
This could cause an infinite loop if the AO did not support spdif, but
was falling back to some PCM format. Then this code tried to remove the
last filter, which is a dummy filter for receiving and queuing filter
output. af_remove() simply fails gracefully in this case, so this
happens over and over again.
Fix by explicitly checking whether the filter to remove is a dummy
filter. (af_remove() also fails only if the dummy filters are attempted
to be removed - checking this directly is simpler.)
(cherry picked from commit 0025030cef)
It appears this is the reason coreaudio-exclusive does not work without
explicitly specifying a device, even if the default device maps to
something passthrough-capable.
(cherry picked from commit 7a5f5a8adf)
Instead of always picking a somehow better format over the previous one,
select a format that is equal to or better the requested format, but is
also reasonably close.
Drop the mFormatID comparison - checking the sample format handles this
already.
Make sure to exclude channel counts that can't be used.
(cherry picked from commit fd6809f98a)
Reduces (but likely does not remove) the danger of rounding intermediate
values down to 8 bit. This is important for cscale, or any other
processing that might store raw YUV values in framebuffers.
Fixes#1918.
(cherry picked from commit cf210c4ffc)
If for example the audio settings are set to 5.1 output, but the
hardware does 8 channels natively (HDMI), the reported channel
layout will have 2 dummy channels. To avoid falling back to stereo,
we have to write audio in this format to the device.
(cherry picked from commit 4d8a7e0394)
Some audio APIs explicitly require you to add dummy channels. These are
not rendered, and only exist for the sake of the audio API or hardware
strangeness. At least ALSA, Sndio, and CoreAudio seem to have them.
This commit is preparation for using them with ao_coreaudio.
The result is a bit messy. libavresample/libswresample don't have good
API for this; avresample_set_channel_mapping() is pretty useless.
Although in theory you can use it to add and remove channels, you
can't set the channel counts. So we do the ordering ourselves by making
sure the audio data is planar, and by swapping the plane pointers. This
requires lots of messiness to get the conversions in place. Also, the
input reordering is still done with the "old" method, and doesn't
support padded channels - hopefully this will never be needed. (I tried
to come up with cleaner solutions, but compared to my other attempts,
the final commit is not that bad.)
(cherry picked from commit 06050aed99)
ca_label_to_mp_speaker_id() checked whether the last entry was >= 0, but
actually this condition was never true, and MP_SPEAKER_ID_UNKNOWN0 is
not negative.
(cherry picked from commit eead97f103)
This was a mistake, it should definitely be using the device namespace
rather than the file namespace. As it says in the docs, all pipe names
must start with \\.\pipe\
(cherry picked from commit b6381a0ee3)
This will probably disable this code for Cygwin. I don't know if this
matters, since Cygwin should strictly behave like a Unix anyway.
(cherry picked from commit 3508a3fbd1)
And split the Cocoa and Unix cases. Simplify the Cocoa case slightly by
calling mpv_main directly, instead of passing a function pointer. Also
add a comment explaining why Cocoa needs a special case at all.
(cherry picked from commit 1e7831070f)
This unbreaks compiling command line player and libmpv at the same
time. The problem was that doing so silently disabled the OSX
application thing - but the command line player can not use the
vo_opengl Cocoa backend without it.
The OSX application code is basically dead in libmpv, but it's not
that much code anyway.
If you want a mpv binary that does not create an OSX application
singleton (and creates a menu etc.), you must disable cocoa
completely, as cocoa can't be used anyway in this case.
(cherry picked from commit 19a5b20752)
Somewhat less ifdeffery, higher flexibility. Now there are 3 separate
config file resolvers for 3 platforms (unix, win, osx), and they can
still interact with each other somewhat. For example, OSX for now uses
most of Unix, but adds the OSX bundle path.
This can be extended to resolve very specific platform paths, such as
location of the desktop.
Most of the Unix specific code moves to path-unix.c.
The behavior should be the same - if not, it is likely a bug.
(cherry picked from commit d3a3cfe54c)
(Not sure why it worked without this when I tested the previous
changes.)
Untested, but should be fine. This is equivalent what is done on e.g.
panscan changes.
(cherry picked from commit 94a3a76ee3)
I think this used to be quite important, because the ancient VfW support
in MPlayer used to output flipped frames. This code has been dead in mpv
for quite some time (because VfW decoders were removed, and the --flip
option was dropped too), so get rid of it.
(cherry picked from commit e185887ba0)
Currently, the wayland backend needs extra work to avoid drawing more
often than the wayland frame callback allows. (This is not ideal, but
will be fixed at a later time.)
Unify this with the start_frame callback added for cocoa. Some details
change for the better. For example, if a frame is dropped, and a redraw
is done afterwards, the actually correct frame is redrawn, instead
whatever was in the textures from before the dropped frame.
(cherry picked from commit 0a7abbda6b)
With --idle --force-window, or when started from the bundle, the cocoa
code dropped the first frame. This resulted in a black frame on start
sometimes.
The reason was that the live resizing/redrawing code was invoked, which
simply set skip_swap_buffer to false, blocking redrawing whatever was
going to be rendered next. Normally this is done so that the following
works:
1. vo_opengl draw a frame, releases GL lock
2. live resizing kicks in, redraw the frame
3. vo_opengl wants to call SwapBuffers, drawing a stale buffer
overwritten by the live resizing code
This is solved by setting skip_swap_buffer in 2., and querying it in 3.
Fix this by resetting the skip_swap_buffer at a known good point: when
vo_opengl starts drawing a new frame.
The start_frame function returns bool, so that it can be merged with
is_active in a following commit.
(cherry picked from commit e23e4c7c60)
Commit f1746741de changed the drop
logic to have more slack (drop more frames but less frequent) to prevent
drops due to timing jitter when the clip and screen have similar rates.
However, if the clip has higher rate than the screen (or just higher
playback rate), then that policy hurts smoothness since these "chunked
drops" look worse than one frame drop at a time.
This patch restores the old drop logic when the playback frame rate is
higher than ~5% above the screen refresh rate, and solves this issue.
Fixes#1897
(cherry picked from commit ffcad1a72b)
Also factor the display size initialization into a separate function.
For some reason this seems to work, although setting the background
color using this 1x1 pixel bitmap does not work. I blame the RPI
beign a terrible piece of hardware with even worse drivers.
(cherry picked from commit 6ae66e717f)
This should for now be equivalent; it's merely more explicit and will
be required if we add PCM support.
Note that the property listeners actually tell you what property
exactly changed, but resolving the current listener mess would be too
hard. So check for changes manually.
(cherry picked from commit 382434d45a)
Useful with some of the following commits.
ca_fill_asbd() should behave exactly as before.
Instead of actually implementing the inverse function of ca_fill_asbd(),
just loop over the (small) list of mpv functions and check if any mpv
equivalent to a given ASBD exists.
(cherry picked from commit 32b835c03b)
kAudioFormatFlagIsSignedInteger implicates that it's only used with
integer formats. The mpv internal flag on the other hand signals the
presence of a sign, and this is set on float formats.
Until now, this probably worked fine, because at least AudioUnit is
ignoring the uncorrect flag.
(cherry picked from commit 3295ce48ab)
The previous version of this logic resulted in black crush and incorrect
brightness scaling when combined with limited range (TV) output.
(cherry picked from commit 99439f11ea)
It's weird that this basically adjusts the contrast between luma and
chroma, and not blackness.
This is more in line with the behavior of libswscale, the vdpau
"procamp" (which mpv doesn't use), and Xv.
(cherry picked from commit 0600d378f9)
This code does not know whether the stream supports reconnecting until
STREAM_CTRL_RECONNECT is called. So the message should be printed after
it. To avoid that reconnects that succeed on the first try go unnoticed,
print a warning on success.
(cherry picked from commit 38114b6a36)
Right now, the default behavior is to pick the numerically lowest screen
ID that overlaps the window in any way - but this means that mpv will
decide to pick an ICC profile in a pretty arbitrary way even if the
window only overlaps another screen by a single pixel.
The new behavior is to query it based on the center of the window
instead.
(cherry picked from commit daf4334697)
Whether this is correct is unknown. This change tripples the latency
from ~15ms to ~45ms.
XBMC does this, VLC does not from what I could see.
(cherry picked from commit 5f86fad2f0)
Should help with debugging, and might be slightly more userfriendly.
Note that this is called manually in multiple entry-points, instead of
the functions doing the actual work (like mp_remove_track()). This is
done so that exiting the player or calling the sub_reload command won't
print redundant in-between states.
(cherry picked from commit 0c0c8cd44e)
During seeking, and there is momemtarily no new data available yet, the
player will display the seek target as current time. Clamp this time to
the known time range as implied by the start time and the duration of
the file.
This improves behavior especially when seeking in audio files, for which
this for some reason triggers rather often. There were some users
complaining about this.
This makes behavior worse for files with timestamp resets, or
incorrectly reported duration. (The latter is relatively common,
e.g. libavformat shortcomings, or incomplete files.)
(cherry picked from commit 0ff93a8357)
We always want to prefer upmix to downmix, as long as it makes sense.
Even if the upmix is not "perfect" (not just adding channels), we want
to prefer the upmix.
Cleanup for commit d3c7fd9d.
(cherry picked from commit c4aa136155)
We escape only characters below 32, plus " and \. UTF-8 should be apssed
through verbatim. Since char can be signed (and usually is), the check
broke and happened to escape UTF-8 encoded bytes too. This broke UTF-8
completely.
Note that we don't check for broken or invalid UTF-8, such as described
both in the client API and IPC docs.
Fixes#1874.
(cherry picked from commit f77e3cbf0c)
This will change the format option to "bestaudio/best" instead of
passing the "-x" argument to yt-dl.
Prevents the video still being downloaded in the new mpv versions where
the yt-dl format is set to "best" by default.
(cherry picked from commit 5555f6fc2c)
As indicated by the added test. In this case, fallback and downmix have
the same score, but fallback happens to give better results. So prefer
fallback over downmix.
(This is probably not a correct solution.)
(cherry picked from commit d3c7fd9d7c)
This could make the player crash on exit if the "sub_reload" command was
used successfully. the reason was that the mpctx->sources array could
have dangling pointers to the unloaded demuxers.
Also fix a memory leak by actually always freeing the per-stream
subtitle decoders (which are a hack to make ordered chapters behave
better).
(cherry picked from commit 4d2ed847ce)
If the --ytdl-format option is not used, force the "best" format. Do
this because youtube-dl is going to change its default format to one
that will trigger the (partially broken) DASH support in our own code.
Fixes#1867.
(cherry picked from commit 95a0488ac2)
vo_opengl (or gl_hwdec_vdpau.c to be specific) calls
mp_vdpau_mixer_render() with video_rect=NULL, which means to use the
full surface. This is incorrect if the surface is actually cropped, as
it can happen with h264. In this case, it was rendering the parts
outside of the image.
Fix it by making this case use the cropped size instead.
Alternative fix for PR #1863.
(cherry picked from commit a5ed6e49bf)
* master: (87 commits)
manpage: move --autosync description
player: add --window-scale option
player: flush decoder even if cover art is decoded
player: don't show A/V desync message in non-sense situations
w32_common: add more rounded-down frame rates
w32_common: use the current monitor's refresh rate
dxva2: fix broken build with gcc 5.1
terminal: printf() is not signal-safe
man: fix PDF build
DOCS/mplayer-changes: Eleborate on joystick input
osc: add nil check for element.eventresponder
mp_image: remove some unused interlacing flags
vf_vapoursynth: update _FieldBased semantics
options: remove unneeded hack from command line parser
manpage: document ff-index sub-property
demux_mkv: limit timestamp fixing to 1ms max
demux_mkv: attempt to fix rounded timestamps
demux_mkv: move global options to the demuxer
demux_mkv: better seeking after video end
lua: add utils.format_json() function
...
This was in the "Window" section. It has absolutely nothing to do with
windows. Move it to the "Miscellaneous" section instead. The "--mc"
option, which has a similar function, was already there.
Fixes PNG cover art not showing up immediately (for example when running
with --pause).
libavformat exports embedded cover art as a single packet. For example,
a PNG attachment simply contains the PNG image, which can be sent to the
decoder. Normally you would expect that the PNG decoder would return 1
frame for 1 packet, without any delays. But this stopped working, and it
incurs a 1 frame delay.
This is perfectly legal (even if unexpected), so let our code feed the
decoder packets until we get something back. (In theory feeding the
packet instead of a real flush packet is still somewhat questionable.)
last_av_difference can be MP_NOPTS_VALUE under certain circumstances
(like no video timestamp yet). This triggered the desync message,
because fabs(MP_NOPTS_VALUE) is quite a large value. We don't want to
show a message in this situation.
The build failed because rst2pdf apparently has problems with
page breaks. In this case, the link to the ALSA upmix guide was
causing a page break in an admonition block. My guess is that
rst2pdf screws up when it can’t fill at least one line of text
following a page break, so I worked around this by making that
paragraph a little longer. Seems to do the trick.
I also shortened the URL using GitHub’s service because it was
causing some rather unsightly formatting in the manpage output.
Maybe we should just build HTML instead of a PDF.
Since joystick support was removed and is a difference from mplayer, it
should be included in the document with the mplayer changes.
It will help new users who were using mplayer's joystick support to
seek alternatives when switching to mpv. It will also be helpful for
people that had problems with the joystick support in mplayer (for
example, by incorrectly recognizing other input devices as joystick)
to know that those problems won't persist in mpv.
MP_IMGFIELD_TOP/MP_IMGFIELD_BOTTOM were completely unused, and
MP_IMGFIELD_ORDERED was always set (even though vf_vdpaupp.c strangely
checked for the latter).
This was traditionally needed to silence terminal output from errors
during command line parsing preparsing. Preparsing is done so that
options controlling the terminal and config files are parsed and applied
first, with a second command line parsing pass applying all other
options, _and_ printing error messages for the preparsed ones.
But the hack silencing log output during the preparse pass is actually
not needed anymore, since the terminal is enabled only after preparsing
is finished. update_logging() in main.c does this.
So as long as update_logging() is called before
m_config_preparse_command_line(), this will work.
The options don't change, but they're now declared and used privately by
demux_mkv.c. This also brings with it a minor refactor of the subpreroll
seek handling - merge the code from playloop.c into demux_mkv.c. The
change in demux.c is pretty much equivalent as well.
This change allows forward seeking even if there are no more video
keyframes in forward direction. This helps with files that e.g. encode
cover art as a single video frame (within a _real_ video stream - ffmpeg
seems to like to produce such files). Seeking backwards will still jump
to the nearest video frame, so this improvement has limited use.
The old code didn't do this because of the logic the min_diff variable
followed. Instead of somehow using the timestamp of the last packet read
for min_diff, use the first index entry for it. This actually makes it
fall back to the first/last index entry as the (removed) comment claims.
Note that last_pts is basically random at this point (because the
demuxer can be far ahead of playback position), so this didn't make
sense in the first place.
An approximate measure to make it exit possibly slightly earlier.
Relatively speaking, some time will pass between cancellation and
the cache actually being requested to exit, so it's good if the
cache returns EOF immediately.
The caller can check for cache interruption instead. There's no need to
define special return values and such. It would be rather hard to make
waiting for the condition and stream cancellation atomic too (and
pointless, since the underlying stream will also be "cancelled" and exit
early), so nothing about cancellation being a separate call will change.
This put some effort into distinguishing between two messages to print -
all worthless. Even more so, this kept printing the message, which
doesn't feel overly useful either. (The message will be printed
repeatedly anyway if network recovers for a while and then gets stuck
again.)
All in all, the demuxer cache triggering the buffering state does a
better job here. But don't remove it completely, since knowing that the
network did nothing for a relatively short time is still useful.
It's annoying and unnecessary. They can be manually loaded if really
needed (for things like previewing).
Also remove the #if. It was for suppressing warnings, and we don't need
to be so careful about this in the relatively obscure encoding mode.
Approximate time of video buffered in the demuxer, in seconds. Same as
`demuxer-cache-duration` but returns the last timestamp of bufferred
data in demuxer.
Signed-off-by: wm4 <wm4@nowhere>
If user switched terminals frantically, mpv could get SIGUSR1 twice in a
row, which, up until now, resulted in destroying CRTC twice. This caused
it to segfault. After this fix, double SIGUSR1 should be ignored.
This will be used in the following commit, which adds screenshot_raw.
The reasoning is that this will be better for binding scripting
languages.
One could special-case the screenshot_raw commit and define fixed
semantics for passing through a pointer using the current API, like
formatting a pointer as string. But that would be ridiculous and
unclean.
On EOF, this stopped reporting the actual cache duration, and just
signalled unknown duration. Fix this and keep reporting whatever is left
in the packet queue.
This reverts commit 5438a8b3. The commit doesn't give a good explanation
as to why it is needed, but I guess it was because the reporting was
imperfect (it switched between unknown or 0, and the correct duration).
This also removes a line added in commit 848546f2. The line is
ds->active = false;
The "active" flag basically says that data from this stream is actively
needed, and it's used to calculate the minimum data that can actually be
played (approximately). If this were ignored, a sparse subtitle stream
would set the cache duration to 0s. The commit message adding the line
says "actually does nothing, but in theory it's cleaner". Well, screw
it.
Vapoursynth made an incompatible API change: clips with unknown length
are not supported anymore. In fact, Vapoursynth abort()s the program
(which by the way invalidate all of its claims of API/ABI stability).
So add some nonsense to make it work again.
Remove the old implementation for these properties. It was never very
good, often returned very innaccurate values or just 0, and was static
even if the source was variable bitrate. Replace it with the
implementation of "packet-video-bitrate". Mark the "packet-..."
properties as deprecated. (The effective difference is different
formatting, and returning the raw value in bits instead of kilobits.)
Also extend the documentation a little.
It appears at least some decoders (sipr?) need the
AVCodecContext.bit_rate field set, so this one is still passed through.
This prevents the machine from going to sleep while a video-only stream
is playing. When audio is playing, the audio stack should make this
request for us.
Useful for dealing with libavfilter's terrible graph syntax.
Not strictly backwards compatible (for example "[a[b]" fails now - the
"[" within the quote is interpreted now). But hopefully it's obscure
enough not to warrant any kind of compatibility hacks.
Logging was meant to be silenced only when user uses connector
auto-detection feature. If user supplies connector ID manually, he
should see exact reason why connecting to this specific connector
failed.
If a directory is encountered, replace it with its contents in the
internal playlist.
This is messed into demux_playlist.c, because why not. STREAMTYPE_DIR
could be avoided by unconditonally trying opendir() in demux_playlist.c,
but it seems nicer not to do weird things like calling it on real files.
This does not work on Windows, because msvcrt is retarded.
It's entirely useless, especially now that vo.c handles screenshots in a
generic way, and requires no special VO support. There are some
potential weird use-cases, but actually I've never seen it being used.
The old behavior does not make too much sense after all. If you don't
want to file to be overwritten, the user can check this manually.
This is a change in behavior - let's hope nobody actually relied on it.
libavcodec makes it impossible to distinguish dropped frames (requested
with AVCodecContext.skip_frame), and cases when the decoder simply does
not return a frame by default (such as with VP9, which has invisible
reference frames).
This confuses users when decoding VP9 video. It's basically a cosmetic
issue, so just paint it over by ignoring them if framedropping is
disabled.
It appears youtube-dl sometimes asks for a password on stdin. This won't
work, because mpv already uses the terminal.
(I wonder if this could be simpler, like simply closing FD 0, but let's
not. The FD would be reused by something random.)
We already use 2 screensaver APIs when attempting to disable the
screensaver: XResetScreenSaver() (from xlib) and XScreenSaverSuspend
(from the X11 Screen Saver extension). None of these actually work.
On modern desktop Linux, we are expected to make dbus calls using some
freedesktop-defined protocol (and possibly we'd have to fallback to a
Gnome specific one). At least xscreensaver doesn't respect the "old"
APIs either.
Solve this by running the xdg-screensaver script. It's a terrible, ugly
piece of shit (just read the script if you disagree), but at least it
appears to work everywhere. It's also simpler than involving various
dbus client libraries.
I hope this can replace the --heartbeat-cmd option, and maybe we could
remove our own DPMS/XSS code too.
It seems this choice was never documented. "always" is actually older
than "yes", so just declare it a compatibility value for "yes". (Also
move it before "always" in the C code to make this clear.)
mp_chmap_from_channels_alsa() doesn't always succeed - there are a bunch
of channel counts for which no defined ALSA layout exists. Fallback to
stereo in this case. (Normally, this code path shouldn't happen at all.)
When playing cover art, it conceptually reaches EOF as soon as the image
was put on the VO, causing the EOF message to be repeated every time new
audio was decoded. Just silence the message.
This was mapped to a broken command, so it did nothing but printing an
error message. The intended binding (cycling OSD level) doesn't seem to
useful either, so just drop it.
It was ignored before. Passing an argument makes no sense, and might be
mistaken for some form of --vf-del, so complain about it.
This also affects --af-clr and the vf/af commands.
This is optional, but ensures that linking with -Wl,--as-needed does
not drop the MMAL VC driver. The driver normally "registers" itself
in the library constructor, but since no symbols are explicitly
referenced, the linker could remove it with as-needed enabled.
I tried to find that option by searching for terms like “cover art”
and got nothing. I imagine most users would look for similar terms.
Hope this helps.
* master: (39 commits)
Update license headers
af_lavrresample: minor simplification
af_lavrresample: allow resetting output sample format
audio/filter: fully renegotiate audio formats on every reconfig
DOCS/waf: Edit --enable-feature sentences
vo_opengl: slightly simplify check_gl_features()
vo_opengl: unify blend-subtitles-res and blend-subtitles
vo_opengl: fix blend-subtitles-res=video & anamorphic video
manpage: mention how pseudo-gui mode is enabled on win32
main-fn-win: fix handle validity check for XP
win32: use pseudo-gui profile when started without stdio
win32: only attach to the console from mpv.com
win32: use a platform-specific unicode entry-point
vo_opengl: add blend-subtitles-res
manpage: document pseudo-gui stuff
mp_image: remove redundant flags field
mp_image: remove redundant chroma_x/y_shift fields
mp_image: remove redundant plane_w/h fields
player: use config parser for setting up pseudo-gui profile
player: change pseudo-gui settings
...
It could happen that a lavrresample filter would keep its old output
format when the decoder changed its output format. This simply happened
because the output format was never reset.
Normally, this was not an issue, because lavrresample filters only
inserted for format conversion were removed on format changes. But if
--no-audio-pitch-correction is set and playback speed is changed, then
there is a "permanent" lavrresample filter in the filter chain, which
shows this behavior.
Fix by explicitly resetting output formats for all filters which support
it.
Note: this can crash with libswresample in some cases. I'm not sure if
this is mpv's fault or libswresample's, but since it works with
libavresample, I'm going to assume it's not our's.
The gender specific pronoun is changed, since we shouldn't assume the
gender of the user.
The sentence itself is also changed to be more correct in general.
Not sure why this was so roundabout; probably to keep spam down if the
user's OpenGL drivers are crap (but then just don't enable extended
features), or because the "Disabling..." text was so repetitious.
But there doesn't seem to be a good reason after all. Also, this could
already overflow the fixed size disabled[] array. Just print the
messages directly.
Apparently the standard handles can be set to bogus values on XP. Use
GetFileType to check whether they refer to an actual file/pipe/etc. The
logic used by is_valid_handle() is now pretty similar to what the CRT
uses to check for valid stdio handles.
If mpv is started from Explorer or the Start Menu, it will have no
console and no standard IO handles. In this case, it's fairly safe to
enable the pseudo-gui profile.
Previously, mpv.exe used the --terminal option to decide whether to
attach to the parent process's console, which made it impossible to tell
whether mpv would attach to the console before the config files were
parsed. Instead, make mpv always attach to the console when launched
from the console wrapper (mpv.com) and never attach otherwise. This will
be useful for the next commit, which will use the presence of the
console to decide whether to use the pseudo-gui profile.
This change should also be an improvement in behavior. The old code
would attach to the parent process's console, regardless of whether it
was mpv.com or some other program like cmd.exe. This could be confusing,
since mpv.exe is marked as a Windows GUI program and shouldn't write
text to its parent process's console when launched directly. (See #768.)
Visual Studio does something similar with its devenv.com wrapper.
devenv.exe only attaches to the console when launched from devenv.com.
Add a platform-specific entry-point for Windows. This will allow some
platform-specific initialization to be added without the need for ugly
ifdeffery in main.c.
As an immediate advantage, mpv can now use a unicode entry-point and
convert the command line arguments to UTF-8 before passing them to
mpv_main, so osdep_preinit can be simplified a little bit.
Because gcc (and clang) is a goddamn PITA and unnecessarily warns if
the universal initializer for structs is used (like mp_image x = {})
and the first member of the struct is also a struct, move the w/h
fields to the top.
Remove --keep-open. Switch to --idle=once. This effectively makes the
player quit after end of playback, but still shows the idle screen if it
was started with no files.
Starting the command line player with --no-terminal, the terminal was
sitll initialized. This happened because update_logging() used the
option value before the options were parsed. Fix by moving down the
initialization to before the point where it's actually needed.
They are redundant. They were used by draw_bmp.c only, and only in a
special code path that 1. used fixed image formats, and 2. had image
sized perfectly aligned to chroma boundaries (so computing the chroma
width/height is trivial).
This could help in cases where the DWM (Windows desktop compositor) adds another
layer of bufferring and therefore the SwapBuffers timing could get messed up.
Signed-off-by: wm4 <wm4@nowhere>
on my windows system this allows smoothmotion to work perfectly also in windowed
mode. There's no real right or wrong here, with the the only goal being to
always hit the next vsync. however, on cases where vsync timing is jittery (as
could happen with DWM), this patch tries to aim to the middle of the vsync cycle
to get as least affected as possible by such jitter.
adds 1 vsync interval "slack" before deciding to drop the first frame. it should
help on cases of timing jitter (sleep duration, container timestamps, compositor
vsync timing, etc). once the drop threshold has been crossed, it will keep
dropping until perfect timing alignment. this prevents crossing the drop
threshold back and forth repeatedly and therefore more resilient to frame drops
When setting options like --no-video, ytdl_hook adds the "-x" argument
to youtube-dl, so that bandwith is saved by not downloading the video on
some sites.
Increase the default queue size. This helps with "missed" frames due to
the asynchronous nature of the API. All the other VOs are synchronous,
so if rendering and displaying takes a while, the common code in vo.c
will be blocked until it can continue. But with opengl-cb, vo.c might
immediately push the next ready frame, which causes the current frame
to be dropped _if_ it wasn't rendered yet.
One could fix this by making vo.c wait a while (until the API user calls
the render function, which pulls the frame). But setting the default
queue size to 2 seems much simpler: instead of dropping the frame, it
will be pushed to the API user once the next renderer call finishes.
(This is still a bit strange, and will hopefully be cleaned up when
video scheduling is redone, but for now this appears to deliver
relatively good results.)
All commits are bugfixes anyway, so just taking everything.
* master:
sub: don't truncate timestamps to int
sub: unprefer libavcodec's MicroDVD converter
manpage: lua: clarify timer usage
audio: automatically deatch filters if spdif prevents their use
audio: change a detail about filter insertion
ao_alsa: change log output
audio: avoid one more redundant audio filter reinit
ao_coreaudio: do not error if retrieving info for verbose mode fails
travis: re-enable OSX
video/out: fix screenshot image formats
vdpau: don't use a transparent image format for screenshots
screenshot: use GPU readback only for hardware decoded images
manpage: remove confusing statement
screenshots: select best image format the encoder supports
Prefer the builtin one again. libavcodec still uses the ASS packet
format that uses inline timestamps, so the packet timestamps are
ignored. This again leads to additional rounding of timestamps, because
the ASS storage format only has 10ms resolution (instead of 1ms
resolution like libass). This again can lead to unintentional overlaps
when converting subtitles. The internal MicroDVD converter avoids this,
because it always uses packet timestamps.
This seems to come up often. I guess '.' vs. ':' for Lua calls is
confusing, and this part of the scripting API is the only one which
requires using it.
The af_add() function has a problem: if the inserted filter returns
AF_DETACH during init, the function will have a dangling pointer. Until
now this was avoided by making sure none of the used filters actually
return AF_DETACH, but it's getting infeasible.
Solve this by requiring passing an unique label to af_add(), which is
then used instead of the pointer.
Silence the usually user-visible warning about unsupported channel maps.
This might be an ALSA bug, but ALSA will never fix this behavior anyway.
(Or maybe it's a feature.)
Log some other information that might be useful.
Only reinit filters if it's actually needed. This is also slightly
easier to understand: if you look at the code, it should now be more
obvious why a reinit is needed (hopefully).
The message log level shouldn't get to decide whether something fails
or not. So replace the fatal error check on the verbose output code
path with a warning.
Avoids a confusing message printed by the vdpau code when taking a
screenshot while using software decoding (because obviously GPU readback
won't work on normal in-memory video frames).
There still might be FFmpeg demuxers which mess up if audio is disabled
(like it happened to the FLV demuxer), but these are bugs and shouldn't
happen.
This matters for png screenshots. We used to hardcode rgb24, but
libavformat's png encoder can do much more. Use the image format list
provided by the encoder, and select the best format from it (according
to the source format).
As a consequence, rgb48 (i.e. 16 bit per component) will be selected if
the source format is e.g. 10 bit yuv. This happens in accordance to
FFmpeg's avcodec_find_best_pix_fmt_of_list() function, which assumes
that 16 bit rgb should be preferred for 10 bit yuv.
This also causes it to print this message in this case:
[ffmpeg] swscaler: full chroma interpolation for destination format 'rgb48be' not yet implemented
I'm not 100% sure whether this is a problem.
Use texture-from-pixmap instead of vaapi's "native" GLX support.
Apparently the latter is unused by other projects. Possibly it's broken
due that, and Intel's inability to provide anything non-broken in
relation to video.
The new code basically uses the X11 output method on a in-memory pixmap,
and maps this pixmap as texture using standard GLX mechanisms. This
requires a lot of X11 and GLX boilerplate, so the code grows. (I don't
know why libva's GLX interop doesn't just do the same under the hood,
instead of bothering the world with their broken/unmaintained "old"
method, whatever it did. I suspect that Intel programmers are just
genuine sadists.)
This change was suggested in issue #1765.
The old GLX support is removed, as it's redundant and broken anyway.
One remaining issue is that the first vaPutSurface() call fails with an
unknown error. It returns -1, which is pretty strange, because vaapi
error codes are normally positive. It happened with the old GLX code
too, but does not happen with vo_vaapi. I couldn't find out why.
We handle picking out font attachments by mime type ourselves in a
higher level, so we really just want to use the mimetype. Also, Matroska
is currently the only code in libavformat which uses the fonts at all,
and we can drop use of the codec IDs completely.
This is a peculiar filter I stumbled upon while playing around with
windows, it removes aliasing almost completely while not ringing at all.
The downside is that it's quite blurry, but at high resolutions it's not
so noticeable.
This merges all of the scaler-related options into a single
configuration struct, and also cleans up the way they're passed through
the code. (For example, the scaler index is no longer threaded through
pass_sample, just the scaler configuration itself, and there's no longer
duplication of the params etc.)
In addition, this commit makes scale-down more principled, and turns it
into a scaler in its own right - so there's no longer an ugly separation
between scale and scale-down in the code.
Finally, the radius stuff has been made more proper - filters always
have a radius now (there's no more radius -1), and get a new .resizable
attribute instead for when it's tunable.
User-visible changes:
1. scale-down has been renamed dscale and now has its own set of config
options (dscale-param1, dscale-radius) etc., instead of reusing
scale-param1 (which was arguably a bug).
2. The default radius is no longer fixed at 3, but instead uses that
filter's preferred radius by default. (Scalers with a default radius
other than 3 include sinc, gaussian, box and triangle)
3. scale-radius etc. now goes down to 0.5, rather than 1.0. 0.5 is the
smallest radius that theoretically makes sense, and indeed it's used
by at least one filter (nearest).
Apart from that, it should just be internal changes only.
Note that this sets up for the refactor discussed in #1720, which would
be to merge scaler and window configurations (include parameters etc.)
into a single, simplified string. In the code, this would now basically
just mean getting rid of all the OPT_FLOATRANGE etc. lines related to
scalers and replacing them by a single function that parses a string and
updates the struct scaler_config as appropriate.
This makes the core much more elegant, reusable, reconfigurable and also
allows us to more easily add aliases for specific configurations.
Furthermore, this lets us apply a generic blur factor / window function
to arbitrary filters, so we can finally "mix and match" in order to
fine-tune windowing functions.
A few notes are in order:
1. The current system for configuring scalers is ugly and rapidly
getting unwieldy. I modified the man page to make it a bit more
bearable, but long-term we have to do something about it; especially
since..
2. There's currently no way to affect the blur factor or parameters of
the window functions themselves. For example, I can't actually
fine-tune the kaiser window's param1, since there's simply no way to
do so in the current API - even though filter_kernels.c supports it
just fine!
3. This removes some lesser used filters (especially those which are
purely window functions to begin with). If anybody asks, you can get
eg. the old behavior of scale=hanning by using
scale=box:scale-window=hanning:scale-radius=1 (and yes, the result is
just as terrible as that sounds - which is why nobody should have
been using them in the first place).
4. This changes the semantics of the "triangle" scaler slightly - it now
has an arbitrary radius. This can possibly produce weird results for
people who were previously using scale-down=triangle, especially if
in combination with scale-radius (for the usual upscaling). The
correct fix for this is to use scale-down=bilinear_slow instead,
which is an alias for triangle at radius 1.
In regards to the last point, in future I want to make it so that
filters have a filter-specific "preferred radius" (for the ones that
are arbitrarily tunable), once the configuration system for filters has
been redesigned (in particular in a way that will let us separate scale
and scale-down cleanly). That way, "triangle" can simply have the
preferred radius of 1 by default, while still being tunable. (Rather
than the default radius being hard-coded to 3 always)
The format doesn't change. Some details are different, though. For
example, it will now accept option values with spaces even if they're
not quoted. (I see no reason why the user should be forced to add
quotes.)
The code is now smaller and should be much easier to extend. It also
can load config from in-memory buffers, which might be helpful in the
future.
read_file() should eventually be replaced with stream_read_complete().
But since the latter function may access options under various
circumstances, and also needs access to the mpv_global struct, there
is a separate implementation for now.
* unify passthrough and pcm exclusive mode format setting/testing
* set passthrough format parameters correctly
* support all of mpv's existing passthrough formats
* automatically test passthrough with exclusive mode and enable
exclusive if it succeeds, even if it was not explictly requested.
this obviates the need for --ao=wasapi,wasapi=exclusive
* if passthrough fails (such as the device doesn't support the
format), fallback to either exclusive pcm or shared mode depending
on what the user specified. Right now this isn't very useful as
it still fails due to the decoder path remainin stuck on spdif.
fixes#1742
Use OPT_CHOICE_C() instead of the custom parser. The functionality is
pretty much equivalent.
(On a side note, it seems --video-stereo-mode can't be removed, because
it controls whether to "reduce" stereo video to mono, which is also the
default. In fact I'm not sure how this should be handled at all.)
The standard say:
NoDisplay means "this application exists, but don't display it in the menus".
Now that --force-window --idle is enabled, we can remove it.
Closes#1755.
libswresample doesn't normalize when remixing to a float format. This
will cause clipping due to float samples being out of the allowed range.
Fortunately this extremely bad default can be changed.
This does not happen with libavresample: it normalizes by default.
Fixes#1752.
This can be set to select a number of default settings that help mpv
pretend that it has a GUI.
I haven't decided yet whether I really want to use the profile mechanism
for this. There are a number of weird details that are not so easy to
handle with profiles, such as disabling pseudo-gui mode again (you can't
unset profiles directly). So this might change. But for now it will do.
There also should be a better way to store builtin profiles.
Unfortunately, the old crappy MPlayer config file parser needs on-disk
files, so just use a bunch of function calls for now.
It simply doesn't work, and is hard to make work. Lua 5.3 is a different
language from 5.1 and 5.2, and is different enough to make adding
support a major issue. Most importantly, 5.3 introduced integer types,
which completely mess up any code which deals with numbers.
I tried to make this a compile time check, but failed. Still at least
try to avoid selecting the 5.3 pkg-config package when the generic "lua"
name is used (why can't Lua upstream just provide an official .pc
file...). Maybe this actually covers all cases.
Fixes#1729 (kind of).
Per-app volume would change the volume across all instances of the same
application, while a private volume control (HAS_PER_APP_VOLUME)
obviously should influence only one instance/audio stream only.
Take advantage of the fact that list_devs is called with a
hotplug_inited ao. Also eliminate unnecessary nested function
abstraction of hotplug_(un)init and list_devs. However, keep list_devs
in ao_wasapi_utils.c since it uses the private functions get_device_id,
get_device_name and exposing these would require including headers for
IMMDevice in ao_wasapi_utils.h.
Create a second copy of the change_notify structure for the hotplug
ao. change_notify->is_hotplug distinguishes the hotplug version from
the regular one monitoring the currently playing ao. Also make the
change notification less verbose now that there might be two of them around.
Rarely used and essentially useless. The only VO for which this was
implemented correctly and for which this did anything was vo_xv, but you
shouldn't use vo_xv anyway (plus it support BT.601 only, plus a vendor
specific extension for BT.709, whose presence this function essentially
reported - use xvinfo instead).
Remove the colorspace-related top-level options, add them to vf_format.
They are rather obscure and not needed often, so it's better to get them
out of the way. In particular, this gets rid of the semi-complicated
logic in command.c (most of which was needed for OSD display and the
direct feedback from the VO). It removes the duplicated color-related
name mappings.
This removes the ability to write the colormatrix and related
properties. Since filters can be changed at runtime, there's no loss of
functionality, except that you can't cycle automatically through the
color constants anymore (but who needs to do this).
This also changes the type of the mp_csp_names and related variables, so
they can directly be used with OPT_CHOICE. This probably ended up a bit
awkward, for the sake of not adding a new option type which would have
used the previous format.
It was "by design" possible to make mpv crash if the parameters didn't
make enough sense, like "format=rgb24:yuv420p". While forcing the format
has some minor (rather questionable) use for debugging, allowing it to
crash is just stupid.
It was already accidentally used unconditionally by command.c.
Apparently this worked well for us, so don't change anything about,
but should it be unavailable, fail at configure time instead of compile
time.
This requires FFmpeg git master for accelerated hardware decoding.
Keep in mind that FFmpeg must be compiled with --enable-mmal. Libav
will also work.
Most things work. Screenshots don't work with accelerated/opaque
decoding (except using full window screenshot mode). Subtitles are
very slow - even simple but huge overlays can cause frame drops.
This always uses fullscreen mode. It uses dispmanx and mmal directly,
and there are no window managers or anything on this level.
vo_opengl also kind of works, but is pretty useless and slow. It can't
use opaque hardware decoding (copy back can be used by forcing the
option --vd=lavc:h264_mmal). Keep in mind that the dispmanx backend
is preferred over the X11 ones in case you're trying on X11; but X11
is even more useless on RPI.
This doesn't correctly reject extended h264 profiles and thus doesn't
fallback to software decoding. The hw supports only up to the high
profile, and will e.g. return garbage for Hi10P video.
This sets a precedent of enabling hw decoding by default, but only
if RPI support is compiled (which most hopefully it will be disabled
on desktop Linux platforms). While it's more or less required to use
hw decoding on the weak RPI, it causes more problems than it solves
on real platforms (Linux has the Intel GPU problem, OSX still has
some cases with broken decoding.) So I can live with this compromise
of having different defaults depending on the platform.
Raspberry Pi 2 is required. This wasn't tested on the original RPI,
though at least decoding itself seems to work (but full playback was
not tested).
There's actually no reason why we should assert. It's unexpected and
"should" not happen, but actually there are several ways to make it
happen.
Still, add a check m_config_get_co(), to avoid matching pseudo-entries
with no name.
Currently, the code just skipped CMS completely. This commit treats them
as sRGB by default, instead.
This also refactors much of the color management code to make it more
generalized and re-usable.
Mktemp on BSDs (at least on OpenBSD) don't have "--tmpdir" option.
Set temporary directory to /tmp by default. Use of ${TMPDIR:=/tmp} allows
overriding temporary directory via environment if needed. (And is indeed needed
in OpenBSD ports infrastructure.)
Signed-off-by: wm4 <wm4@nowhere>
More clearly separate the exclusive and shared mode format discovery.
Make the exclusive mode search more systematic in particular about
channel maps (i.e., use chmap_sel). Assume that the same sample format
/ sample rates work for all channels to narrow the search space.
The code actually uses blocking mode, so opening sound device in non-blocking
mode results in choppy sound. Also, inflating the buffer isn't necessary in
blocking mode, so the function may simply return without doing anything.
This is a minor precaution, because in some cases the number of shader
programs can already hit 10. (chroma merging + separated cscale/scale +
+ sub blending (interpolated) + sub blending (non-interpolated)
+ output (interpolated) + output (non-interpolated) + OSD)
This has a number of user-visible changes:
1. A new flag blend-subtitles (default on for opengl-hq) to control this
behavior.
2. The OSD itself will not be color managed or affected by
gamma controls. To get subtitle CMS/gamma, blend-subtitles must be
used.
3. When enabled, this will make subtitles be cleanly interpolated by
:interpolation, and also dithered etc. (just like the normal output).
Signed-off-by: wm4 <wm4@nowhere>
Bilinear scaling is not a suitable default for something named "hq"; the
whole reason this was done in the past was because cscale used to be
obscenely slow. This is no longer the case, with cscale being nearly
free.
Handling this perfectly with VapourSynth is probably not possible: you
either need to tell it the total number of input frames in advance, or
deliver an infinite stream. With playback, EOF can happen at an
unpredictable point, for which the VapourSynth API has no mechanism at
all. We handle EOF by returning an error to the filter, which will the
filter return all pending frame callbacks.
We still can try to handle it approximately: if the filter requests a
frame past EOF, then send it an error. This seems to work relatively
well with filters which don't request future frames.
With the previous commit, we have no need anymore to check a part of an
extension string (for ignoring a prefix). So check the extension string
properly, instead of just using the broken old strstr() method, which
could accidentally ignore prefixes or suffixes. Do this by extending
the check to whether the extension name is properly delimited by spaces
or string start/end.
Instead of somehow looking for the substring "_swap_control" and trying
to several arbitrary function names, do it cleanly. The old approach has
the problem that it's not very exact, and may even load a pointer to a
function which doesn't exist. (Some GL implementations like Mesa return
function pointers even the functions which don't exist, and calling them
crashes.)
I couldn't find any evidence that glXSwapInterval, wglSwapIntervalSGI,
or wglSwapInterval actually exist, so don't include them. They were
carried over from MPlayer times.
To make diagnostics easier, print a warning in verbose mode if the
function could not be loaded.
Drop support for GL_EXT_framebuffer_object. It has 2 problems: semantics
might be slightly different from the "proper" GL_ARB_framebuffer_object
extension (but is likely completely untested), and also our extension
loader is too dumb to load the same group of function pointers
represented by different extensions only once.
This broke with PulseAudio: when changing some audio filters (like for
playback speed), mixer_reinit_audio() was called - and it overwrote the
volume with whatever mpv thought the volume was before. If the volume
was changed externally before and while mpv was running, this would
reset the volume to the old value.
Fixes#1335.
Pretty messy, which is why it wasn't done at first. (Also, I'd really
like to have simpler syntax and semantics for this damn option, but who
knows when this will happen.)
Fixes#1705.
With a recent cleanup, rar support was stuffed into demux_playlist.c
(because "opening" rar files pretty much just lists archive contents and
adds them to a playlist using a special rar:// protocol, which will
actually access the rar file contents).
Since demux_playlist.c is probed _after_ demux_lavf.c (and should/must
be), libavformat was given the chance to detect DTS streams embedded
within the rar file. This is not really what we want, and a regression
what happened before rar listing was moved to demux_playlist.c.
Fix it by moving the rar listing into its own pseudo-demuxer, and let ir
probe before demux_lavf.c.
(Yes, this feature still has users.)
Why did this exist in the first place? Other than being completely
useless, this even caused some regressions in the past. For example,
there was the case of a laptop exposing its accelerometer as joystick
device, which led to extremely fun things due to the default mappings of
axis movement being mapped to seeking.
I suppose those who really want to use their joystick to control a media
player (???) can configure it as mouse device or so.
We've been prefering the libavcodec mp3 decoder for half a year now.
There is likely no benefit at all for using the libmpg123 one. It's just
a maintenance burden, and tricks users into thinking it's a required
dependency.
Precise seeking requires skipping audio, since the demuxer usually
doesn't seek precisely enough. There is a sanity check that prevents
skipping more than 300 seconds of audio. This still fails with very
large mp3s. For example, with a 1GB sized mp3 with Xing headers, entries
will be 4 MB apart on average, and occasionally much more.
Just bump the limit. I'm not even sure why it was added in the first
place; I suppose it's most important for files with real PTS resets.
When not receiving frame callbacks, we should not draw anything to avoid
blocking the OpenGL renderer. We do this by extending gl context api, by
introducing new optional function 'is_active', that indicates whether
OpenGL renderers should draw or not.
This fixes issue #249.
Define frame callback logic in wayland_common.c
As this should be used by opengl renderer as well.
Preferably drawing should be skipped entierly when no frame callbacks
are received. However, for now only swap buffers is skipped.
Like FFmpeg/Libav do. It seems not all code can actually deal with this
situation, so it's better to shift the special-cases to code which needs
it (possibly OSD code; screenshots of 0x0 windows would just fail).
I guess we don't really care whether this particular function succeeds.
If it fails, it must be completely broken anyway and it would not matter
much to us.
"Non-monotonic" isn't even 100% correct; it's missing "strictly" (for
briefness I guess), and also the message is printed if the PTS jumps
forward. So just print something that is likely a bit easier to
understand.
Unlike other VOs, this rendered OSD even while no VO was created
(because the renderer lives as long as the API user wants). Change this,
and refactor the code so that the OSD object is accessible only while
the VO is created.
(There is a short time where the OSD can still be accessed even after VO
destruction - this is not a race condition, though it's inelegant and
unfortunately unavoidable.)
--af=bs2b:help abort()ed because the default value of the "profile"
option is not represented by any choice. Fix it by adding an "unset"
choice. (It's a bit odd because there's already a "default" choice,
which is not default, but I don't care enough about this filter.)
Fixes#1712.
gl_video_set_options() didn't update it, so the default value set on
initialization was used. Fix by always setting the clear color before
the clear command; it's slightly easier to follow too.
Codecs for hardware acceleration are not blacklisted, but whitelisted.
Also, if this emssage is printed, the codec might not have any hardware
acceleration support in the first place.
Trying to handle such video is almost worthless, but it was requested by
at least 2 users.
If there are no timestamps, enable byte seeking by setting
ts_resets_possible. Use the video FPS (wherever it comes from) and the
audio samplerate for timing. The latter was already done by making the
first packet emit DTS=0; remove this again and do it "properly" in a
higher level.
Normally, the size of an mage plane is assumed to be stride*height. But
in theory, if stride is larger than width*bpp, the last line might not
be padded, simply because it's not necessary. FFmpeg's or mpv's image
allocators always guarantee that this padding exists (it wastes some
insignificant memory for avoiding such subtle issues), but some other
libraries might not.
I suspect one such case might be Xv via vo_xv (see #1698), although my X
server appears to provide full padding. In any case, it can't harm.
There's literally no reason why these functions have to be inline (they
might be performance critical, but then the function call overhead isn't
going to matter at all).
Uninline them and move them to mp_image.c. Drop the header file and fix
all uses of it.
The libavformat rtmp protocol's "timeout" option has two problems:
1) Unlike all other protocols, it's in seconds and not microseconds
2) It enables "listen" mode, which breaks playback
Make the --network-timeout do nothing in the rtmp case.
Fixes#1704.
For some reason there were two points in the code where it warned
against non-monotonic video PTS. The one in video.c triggered on PTS
going backwards or making large jumps forwards, while dec_video.c
triggered on PTS going backwards or PTS not changing. Merge them into a
single check, which warns against all cases.
There was a somewhat obscure optimization in the OSD and subtitle
rendering path: if only the position of the sub-images changed, and not
the actual image data, uploading of the image data could be skipped. In
theory, this could speed up things like scrolling subtitles.
But it turns out that even in the rare cases subtitles have such scrolls
or axis-aligned movement, modern libass rarely signals this kind of
change. Possibly this is because of sub-pixel handling and such, which
break this.
As such, it's a worthless optimization and just introduces additional
complexity and subtle bugs (especially in cases libass does the
opposite: incorrectly signaling a position change only, which happened
before). Remove this optimization, and rename bitmap_pos_id to
change_id.
To handle seeking correctly, we need to flush the filter. libavfilter
does not support flushing, so we destroy and recreate it. We also need
to handle resume-after-EOF, because the mpv audio code sends an EOF
before and after seeking (the latter happens because the player drains
the filter chain in a generic way, which "causes" EOF).
This played e.g. a 1264x722 file as 1264x720. There was some code which
dropped the aspect ratio if the video (in original resolution) wasn't
scaled by more than 4 pixels. Commit 5f3c3f8c introduced this (although
I'm not really sure what the code replaced by it did).
Just remove this "feature".
We now update uniforms every time, so we should try to reduce the number
of uniforms to avoid performance penalties. (Originally, some caching
was planned, but it looks like it would be too complicated to implement
compared to the expected gains.)
OPT_REPLACED can't specify option values or multiple options. Change to
OPT_REMOVED. Also, target-prim doesn't have an srgb option. BT.709 uses
sRGB primaries, so use it instead.
The default scaling was a slight bit too low, which could cause buffer
underruns in some cases.
This should improve the result when using tscale filters other than
oversample. The oversample case should be unaffected.
This adds extra debugging output for buffer underruns, to help track
down possible queueing issues. It also inverts the numberic output for
tscale=oversample to make more sense, without changing the logic.
This moves the color management code out of pass_render_main (which is
now dedicated solely to up/downscaling and hence renamed pass_scale_main)
and into a new function, which gets called from pass_draw_to_screen
instead.
This makes more sense from a logical standpoint, and also means that we
interpolate in linear RGB, before color management - rather than after
it, which is significantly better for color accuracy and probably also
interpolation quality.
This replaces the old smoothmotion code by a more flexible tscale
option, which essentially allows any scaler to be used for interpolating
frames. (The actual "smoothmotion" scaler which behaves identical to the
old code does not currently exist, but it will be re-added in a later commit)
The only odd thing is that larger filters require a larger queue size
offset, which is currently set dynamically as it introduces some issues
when pausing or framestepping. Filters with a lower radius are not
affected as much, so this is identical to the old smoothmotion if the
smoothmotion interpolator is used.
Also the size is now a simple #define that can easily be changed later.
This is done for smoothmotion, which might want to blend more than 4
frames at once, depending on the setting.
Since the gl_rework merge, this started to print some OpenGL errors when
using vdpau hardware decoding with vo_opengl smoothmotion. This happens
because some hwdec unmap_image call were not paired with a map_image
call. Unlike the old vo_opengl, the new code does not do this out of
convenience (it would be a pain to track this exactly). It was triggered
by smoothmotion, because not every rendered frame has actually a new
input video frame (i.e. no map_image call, but it called unmap_image
anyway).
Solve this by handling unmapping differently in the vdpau code. The next
commit will remove the unmap_image callback completely.
Fixes#1687.
This caused complaints because the fps was basically rounded on
microsecond boundaries in the vsync interval (it seemed convenient to
store only the vsync interval). So store the fps as float too, and let
the "display-fps" property return it directly.
Even the lowest supported GL versions have arrays. This test was for
returning arrays from functions, which didn't work in lower GL versions,
but we don't need it anymore.
Previously, mpv would hide the cursor when the autohide timer expired,
even if the window menu was open. This made it difficult to use the menu
with the mouse.
When handling VOCTRL_SET_CURSOR_VISIBILITY, instead of determining
whether to call SetCursor by checking if the cursor is in the client
area, call it based on the parameters to the last WM_SETCURSOR message.
When the window enters "menu mode," it gets a WM_SETCURSOR message with
HIWORD(lParam) set to 0 to indicate that the cursor shouldn't be set.
I think this is what I alwass missed ever since I found the MPlayer
cache options: a way to enable the cache on local files with the default
settings, whatever they are.
Requested change in behavior.
Note that we set the assumed "infinite" display_fps to 1e6, which
conveniently lets vo_get_vsync_interval() return a dummy value of 1,
which can be easily checked against, and still avoids doing math with
float INFs.
I'm not comfortable with VOCTRL_GET_DISPLAY_FPS being called every
frame.
This requires the VO to set VO_EVENT_WIN_STATE if the FPS could have
changed. At least the X11 backend does this.
This adds stuff related to gamma, linear light, sigmoid, BT.2020-CL,
etc, as well as color management. Also adds a new gamma function (gamma22).
This adds new parameters to configure the CMS settings, in particular
letting us target simple colorspaces without requiring usage of a 3DLUT.
This adds smoothmotion. Mostly working, but it's still sensitive to
timing issues. It's based on an actual queue now, but the queue size
is kept small to avoid larger amounts of latency.
Also makes “upscale before blending” the default strategy.
This is justified because the "render after blending" thing doesn't seme
to work consistently any way (introduces stutter due to the way vsync
timing works, or something), so this behavior is a bit closer to master
and makes pausing/unpausing less weird/jumpy.
This adds the remaining scalers, including bicubic_fast, sharpen3,
sharpen5, polar filters and antiringing. Apparently, sharpen3/5 also
consult scale-param1, which was undocumented in master.
This also implements cropping and chroma transformation, plus
rotation/flipping. These are inherently part of the same logic, although
it's a bit rough around the edges in some case, mainly due to the fallback
code paths (for bilinear scaling without indirection).
The basic idea is to use dynamically generated shaders instead of a
single monolithic file + a ton of ifdefs. Instead of having to setup
every aspect of it separately (like compiling shaders, setting uniforms,
perfoming the actual rendering steps, the GLSL parts), we generate the
GLSL on the fly, and perform the rendering at the same time. The GLSL
is regenerated every frame, but the actual compiled OpenGL-level shaders
are cached, which makes it fast again. Almost all logic can be in a
single place.
The new code is significantly more flexible, which allows us to improve
the code clarity, performance and add more features easily.
This commit is incomplete. It drops almost all previous code, and
readds only the most important things (some of them actually buggy).
The next commit will complete it - it's separate to preserve authorship
information.
Starting to get tired of seeing the full config.h in verbose output
every time. Make it slightly more elegant by outputting the list of
satisfied dependencies instead.
Instead of refusing to set properties like "fullscreen" if no VO was
created, always allow it. So if no VO is created, setting the property
merely changes the options (and will be applied once the VO is created).
This is consistent with similar behavior changes to some other
properties.
Improves the behavior reported in #1676.
Also, we shouldn't check the config_ok variable - the VO should do this.
mpctx->filename is obviously not set if no file is playing.
When this code was written, it probably couldn't happen, because the
normal screenshot path fails much earlier in idle mode. But you can
still take screenshots in "full window" mode, and recently the
screenshot code was changed to use the "full window" mode if the normal
path does not work.
If you click on a window that doesn't have a focus, a LeaveNotify
followed by a EnterNotify event can be generated. The former will have
mode set to NotifyGrab, the latter to NotifyUngrab. This will make the
player think the mouse left the window, even though this is not the
case. Ignore these and only react to those with mode set to
NotifyNormal.
Probably fixes#1672, and some other strange issues on some WMs.
In commit 5f8b060e I blindly assumed that the packet sizes were in
pseudo-samples, but they were actually in bytes. Oops.
(The effect was that cutting the audio was a bit less precise than it
can be.)
Also remove the packet size from ad_spdif.c; it didn't actually use it,
and simply takes what the spdif "muxer" returns.
When playback is started after seeking or opening a file, we need to
make sure audio and video line up exactly. This is done by cutting or
padding the audio stream to start on the video PTS.
This does not quite work with spdif: audio is compressed data, within a
spdif frame. There is no way to cut the audio "in between" the frames.
Cutting between the frames would just produce broken spdif packets, and
who knows how receivers will react to this (play noise?). But we still
can cut it in frame boundaries.
Unfortunately, we also insert 0 data for "silence" - we probably
shouldn't do this. Chances are the receiver will switch to PCM or so.
But for now this will have to do.
Note that this could be simplified somewhat, as soon as we work with
frames. See previous commit.
Needed for the next commit. This commit should probably be reverted as
soon as we're working with full audio frames internally, instead of
"flat" FIFOs.
mpv_opengl_cb_render() is supposed to clear the screen with the
background color if there's no video... I think.
It didn't do this, because although uninit() requested gl_video_config()
to be called, this didn't happen, because this function checks whether
the VO is set - and it's unsert after uninit() releases the lock.
Also call the user wakeup callback in this situation, so the user
actually redraws immediately.
Instead of maintaining a private ring buffer, use the generic support
for audio APIs with pull callbacks (internally called AO pull API). This
also fixes latency calculations: instead of just returning the
ringbuffer status, the audio playback state is calculated better and
includes interpolation.
The main reason this wasn't done earlier was mid-stream format
switching. The pull API can now handle it (in a way) by destroying and
recreating the AO. This is a bit brutal, but quite simple. It's untested
in this new AO, though. Some details might not be right, like how ot
restores the old format when reloading.
This could mute a digital passthrough stream by writing zeros. All other
volume values did nothing.
The comment about MPlayer dying hasn't been true in mpv for quite a
while. It's even possible that it's fixed in upstream MPlayer. mpv will
print a scary error message when trying to change volume with spdif, and
continue normally.
If we really want to mute by writing zeros, we should do it in a
separate filter. But I'm not overly fascinated by this approach; is it
even guaranteed receivers will not be confused by a stream of zeros?
The main reason to remove this is that it's in the way of further
cleanups.
Basically, the idea behind the vp parameter is broken - I guess the
intention was to enable rendering to a specific subrectangle of the
target framebuffer, but there's nothing to specify the actual target
rectangle (the VO will still clear e.g. the borders between video and
framebuffer borders).
We're not going to keep the current semantics either with the upcoming
rework of vo_opengl, so declare this for broken. Maybe we can introduce
a function later which does this properly.
Seems appropriate, and will probably avoid performance surprises with
scary architectures which don't have trivial implementations for atomic
loads. (Consider that demux_mkv calls this very often now, and
libavformat demuxers and streams did this for a while now.)
Check async abort notification. libavformat already do something
equivalent.
Before this commit, the demuxer could enter resync mode (and print silly
warning messages) when the stream stopped returning data because of an
abort.
Do not rely on the pointed-to argument to be initialized; VOCTRLs are
supposed to completely overwrite them on success (or not to touch them
on failure).
The currently only caller of VOCTRL_GET_WIN_STATE initializes the value
before calling this, so this is merely about correctness and didn't lead
to any actual bugs.
This causes the cache to be enabled with --cache=auto. It was not done
previously because the small cache size 320k actually led to worse
performance. However, with the current default cache size of 25000kb,
caching notably improves performance.
Add filter parameters to VAAPI deinterlacing filter to actually process
bottom fields instead of deinterlacing top field twice.
Signed-off-by: wm4 <wm4@nowhere>
Some UI elements in OS X – like Launchpad and Dock folders – are implemented
using borderless windows in background demonized applications.
When we use these elements, mpv doesn't stop to be the active application, but
the mpv window leaves keyWindow state while we use the OS controls.
This commit just keeps track of window state to update the cursor visibility
accordingly.
Fixes#513
Handle the failure gracefully, instead of exploding and disabling audio.
Just set the speed back to 1.0.
Also remove the AF_DETACH from af_scaletempo. This actually created a
dangling pointer in af_add(), a tricky consequence of af_add()
reconfiguring the filter chain and the newly added filter using
AF_DETACH. Fortunately the AF_DETACH is not needed (and probably never
worked - it comes from MPlayer times, and MPlayer also disables audio
when trying to change speed with spdif).
Always use af_scaletempo if it's inserted, even if the option
--audio-pitch-correction=no is set.
Make sure all filters are reset on speed change. It's conceivable that
dynamic changes to the filter chain at runtime leave filters around
without resetting their speed parameters.
Also move the code to a separate function.
Make MpvEventsView -signalMousePosition a public method so it can be
called without a compiler warning. Previously, the mouse position would
be reported as (0,0) until the cursor was moved.
mpv would attempt to load ICC profiles several times during VO init
even if no window is displayed. This potentially causes it to load
a profile for a different screen than it is going to be displayed
on, thereby invalidating the profile cache and rebuilding the LUT
every single time.
It would not unload a previously loaded profile when the video
window is moved to a display without an installed profile.
Fix these issues and tweak the log messages a little.
A user reported a webm stream that couldn't be played. The issue was
that this stream 1. was on an unseekable HTTP connection, and 2. had a
SeekHead element (wtf?). The code reading the SeekHead marked the
element as unreadable too early: although you can't seek in the stream,
reading the header elements after the SeekHead read them anyway. Marking
them as unreadable only after the normal header reading fixes this.
(The way the failing stream was setup was pretty retarded: inserting
these SeekHead elements makes absolutely no sense for a stream that
cannot be seeked.)
Fixes#1656.
The global unpack function got moved to table.unpack in Lua 5.2, and
it's only available as the global if 5.2 is built with compatibility
enabled (the default). Lua 5.3 does not build with 5.1 compatibility by
default.
Fixes#1648.
The vaapi equalizer have a custom range, and can have a smaller range
than mpv's normalized video equalizer values. The result is that a vaapi
equalizer value can map to multiple mpv values, so changing a mpv value
by 1 can get "stuck". Fix by remember the mpv value, and returning it if
it still corresponds to the vaapi value.
Really fixes#1647.
(Why am I even bothering with this irredeemable crap?)
If the audio decoder was created, but no audio filter chain created yet
(still trying to decode a first audio frame), setting the "speed"
property could explode. It tried to recreate the filter chain, even
though no format was set yet.
This is inconvenient and should not happen.
Using check_statement() with an empty statement just to check for the
header is quite a hack. Fix check_headers() (so it takes a "use"
parameter), and use it for the checks instead.
main() being called with argc==0 is probably possible. Fix by skipping
the program name early. (I already changed and reverted this once, but
this time we make sure that it's less likely to confuse the skipped argv
with main()'s argv by naming it "options".)
Instead of "vaapi", simply by changing the probe order.
"vaapi" uses the GLX GL interop, which has causing us more problems than
it solved.
Unfortunately this leads also to copying if "--hwdec=auto --vo=vaapi" is
used, even though GLX is not involved in this case - but I don't care
enough to make the probe logic cleverer just for this. You can still get
the zero-copy path with --hwdec=vaapi.
This duplicates the logic which FFmpeg's libavcodec uses. The effects
are unknown, though it's somewhat clear that a single thread doesn't
necessarily saturate a single CPU.
(Eventually we should just let FFmpeg auto-init the thread count, but
for now I prefer it this way, so e.g. verbose mode will print the
thread count.)
Move the command line parsing and some other things to the common init
routine shared between command line player and client API. This means
they're using almost exactly the same code now.
The main intended side effect is that the client API will load mpv.conf;
though still only if config loading is enabled.
(The cplayer still avoids creating an extra thread, passes a command
line, and prints an exit status to the terminal. It also has some
different defaults.)
This gets rid of the need for a second (or more) parameters; instead it
can be all in one parameter. The (now) redundant parameter is still
parsed for compatibility, though.
The way the flags make each other conflict is a bit tricky: they have
overlapping bits, and the option parser disallows setting already set
bits.
MPlayer requires numeric values for input command parameters. mplayer2
also did. mpv changed these to choices using symbolic strings a long
time ago, but left numeric choices for compatibility.
If we're caching a stream with unknown size, and we reach EOF, then
consider the EOF position the file size. Typically makes sense when
reading from a pipe or a http connection that did not send a size.
This automatically sets the gamma option depending on lighting conditions
measured from the computer's ambient light sensor.
sRGB – arguably the “sibling” to BT.709 for still images – has a reference
viewing environment defined in its specification (IEC 61966-2-1:1999, see
http://www.color.org/chardata/rgb/srgb.xalter). According to this data, the
assumed ambient illuminance is 64 lux. This is the illuminance where the gamma
that results from ICC color management is correct.
On the other hand, BT.1886 formalizes that the gamma level for dim environments
to be 2.40, and Apple resources (WWDC12: 2012 Session 523: Best practices for
color management) define the BT.1886 dim at 16 lux.
So the logic we apply is:
* >= 64lux -> 1.961 gamma
* =< 16lux -> 2.400 gamma
* 16lux < x < 64lux -> logaritmic rescale of lux to gamma. The human
perception of illuminance roughly follows a logaritmic scale of lux [1].
[1]: https://msdn.microsoft.com/en-us/library/windows/desktop/dd319008%28v=vs.85%29.aspx
This will be pretty useful to let mpv automatically change VO parameters based
on ambient lighting conditions.
The conversion code and polinomial equation from Apple LMU values to Lux is
taken from Firefox: their license, MPL is GPL compatible and allows
relicensing to GPL (MPL is more liberal).
This reverts commit c8f49be919.
Not needed anymore; fixed in all supported FFmpeg releases. Though I
could not test again, because all sample files are gone (oops).
The af_lavrresample commit made compilation fail on Libav 10, so I think
it's time to require somewhat more recent dependencies.
Libav 11 is the latest release, and FFmpeg 2.4 seems to correspond to
Libav 11. So use these.
Also adjust the configure failure message. Instead of (accidentally)
printing the pkg-config versions twice, print the release version
numbers too. This is helpful, because the release version numbers are
completely different from the pkg-config ones.
I will probably remove some compatibility hacks in the following commits
too.
Although the libraries we use for resampling (libavresample and
libswresample) do not support changing sampelrate on the fly, this makes
it easier to make sure no audio buffers are implicitly dropped. In fact,
this commit adds additional code to drain the resampler explicitly.
Changing speed twice without feeding audio in-between made it crash
with libavresample inc ertain cases (libswresample is fine). This is
probably a libavresample bug. Hopefully this will be fixed, and also I
attempted to workaround the situation that crashes it. (It seems to
point in direction of random memory corruption, though.)
Use the (relatively new) libavformat image format probing functionality,
instead of letting demux_mf guess by file extension and MIME type.
The libavformat support is weird, though. Traditionally, it uses an
absolutely terrible hack to detect images by extension, _and_ (which is
the horrible part) will randomly interpret parts of the filename as
specifiers for matching by number. So something like '%03d' will be
interpreted as placeholder for a frame number. The worst part is that
such character sequences can be perfectly valid and common in http URLs.
This is known as "image2" demuxer. The newer support, which probes by
examining the file header, is split into several format-specific
demuxers with names ending in "_pipe". So we check for such a name
suffix. (At this point we're doing fine-grained hacking around ffmpeg
weirdness, so a clean solution is impossible anyway until upstream
changes.)
It was possible to make the player play local files by putting rar://
links into remote playlists, and some other potentially unsafe things.
Redo the handling of it. Now the rar-redirector (the thing in
demux_playlist.c) sets disable_safety, which makes the player open any
playlist entries returned. This is fine, because it redirects to the
same file anyway (just with different selection/interpretation of the
contents). On the other hand, rar:// itself is now considered fully
unsafe, which means that it is ignored if found in normal playlists.
This warning wasn't overly helpful in the past, and warned against
perfectly fine code. But at least with recent gcc versions, this is the
warning that complains about assignments in if expressions (why???), so
we want to enable it.
Also change all the code this warning complains about for no reason.
Semi-important, because --hwdec=dxva2 outputs NV12, and we really don't
want people to end up with the "old" StretchRect method.
Unfortunately, I couldn't actually get it to work. It seems most D3D
drivers (including the wine D3D implementation) reject D3DFMT_A8L8,
and I could not find any other 2-channel 8 bit Direct3D 9 format. It
seems newer D3D APIs have DXGI_FORMAT_R8G8_UNORM, but there's no way
to get it in D3D9.
Still pushing this; maybe it actually works on some drivers.
This time (there are a lot of times), libswscale randomly ignores
brightness/saturation/contrast settings.
Looking at MPlayer code, it appears the return value of
sws_setColorspaceDetails() signals if changing these settings is
supported at all.
(Nevermind that supporting this feature has almost 0 value, and
obviously eats maintenance time.)
When used with mp.get_screen_size(), mp.get_screen_margins() allows a
Lua script to determine what portion of the mpv window actually has the
video in it.
Breaks vo_opengl by default. I'm hot able to fix this myself, because I
have no clue about the overcomplicated color management logic. Also,
whilethis is apparently caused by commit fbacd5, the following commits
all depend on it, so revert them too.
This reverts the following commits:
e141caa97d653b0dd529729c8b3f64fbacd5de31Fixes#1636.
It is also used for initialization in channel-list setup.
Should fix compilation on FreeBSD, and is more correct
since it is used unconditionally.
Reverts 6445648 .
Just use makeFirstResponder on the mpv events view from client code
if you need the built in keyboard events (this is easier for dealing with view
nesting).
This relies on upstream support in lavc, and will hence basically not
work at all. The intent is to get support for writing this information
into ffmpeg's PNG encoders etc.
We have MP_CSP_TRC defined, but it wasn't being used by practically
anything. This commit adds missing conversion logic, adds it to
mp_image, and moves the auto-guessing logic to where it should be, in
mp_image_params_guess_csp (and out of vo_opengl).
Note that this also fixes a minor bug: csp_prim was not being copied
between mp_image structs if the format was not YUV in both cases, but
this is wrong - the primaries are always relevant.
Refactors an older hack, which for some reason used a more complicated
way. This generates the playlist representing the contents of the rar
file in demux_playlist.c. The pseudo-demuxer could easily be separate
from the the playlist parsers (and in fact there's almost no shared
code), but I don't think this obscure feature deserves a separate file.
Sample files created with:
rar a -v20000k -m0 files.rar file1.mkv file1.mkv
A use of NO_STREAM_ID_FILTER was added to the DVB code recently. While I
have no idea what it's needed for, it makes mpv fail to compile on
FreeBSD 10.1. Add it to the dvb configure check.
In the past it happened quite often that flag options (yes/no) were
changed to choice options (yes/no/some more). The problem with this was
that while flag options don't need a parameter, this wasn't the case
with choice options. A hack was introduced to compensate for this:
setting M_OPT_OPTIONAL_PARAM on the option, and an empty string ("") was
added as choice, so that the choice could be used like a flag. So, for
example, "--mute" would set the choice "".
Fix this by 1. not requiring a parameter if there's a "yes" choice, and
2. redirect an empty parameter to "yes". The effect is that a choice
option with the choices ["yes", "no"] is pretty much equivalent to a
flag option.
Now that we have fast stream switching, we can bump these sizes, as the
queues cause no delay in switching anymore.
Of course, the fast stream switching works for mkv and mp4 only. Other
formats will incur a quite terrible delay especially in network mode,
which this commit changes to 10 seconds. Let's see if someone
complains...
The way I interpreted it, it seemed like this was not default behavior
and could be enabled with --audio-pitch-correction - it should be made
clearer that this is actually *the default behavior*.
This is based on pretty much the same (somewhat naive) logic right now.
I'm not convinced that the extra logic that eg. madVR includes is worth
enough to warrant heavily confusing the logic for it.
This shouldn't slow down the logic at all in any sane shader compiler,
and indeed it doesn't on any shader compiler that I tested.
Note that this currently doesn't affect cscale at all, due to the weird
implementation details of that.
Broken drivers are an issue rather often. Maybe this gives the user an
idea that this could be the reason. (We can't dump much more info on a
80x24 terminal.)
Change test_fbo() so that it checks the FBO lazily, and restructure
check_gl_features() to invoke it only if we know that a FBO will be
needed for a certain enabled feature.
This can avoid strange error messages when using --vo=opengl and the
FBO format does not work. It's also less confusing when reading the
verbose log (users might think that the FBO is actually used, etc.).
This can happen with the "no-colorkey" suboption. Then the code in
xv_draw_colorkey() can be run before vo_x11_config_vo_window(), when
vo_gc is not allocated yet.
Fixes#1629.
This option allows the user to pass non-supported options directly to
youtube-dl, such as "--proxy URL", "--username USERNAME" and
'--password PASSWORD".
There is no sanity checking so it's possible to break things (i.e.
if you pass "--version" mpv exits with random JSON error).
Signed-off-by: wm4 <wm4@nowhere>
If the cache is enabled, the demuxer is closed and opened again (because
currently, the cache can not be enabled atfer data was already read).
The call for opening a new demuxer uses the same params struct, which
references the ctx->uids array. But there is a MP_TARRAY_GROW()
invocation somewhere on the way, which can reallocate the ctx->uids
array, making params.uids a dangling pointer.
This issue probably existed for a longer time, probably since 5cd33853
(slightly more obvious since f50b105d).
Again removes some indirections and extra arguments.
Also replace some memcpy/memmoves with assignments. (Assignments became
possible only later, when reference UIDs were turned into a struct.)
Should behave about the same, but reduces code some duplication with
seeking and reading a header element pointed to by a SeekHead. It also
makes behavior with incomplete files slightly better.
Instead of rendering and upscaling each video frame on every vsync, this
version of the algorithm only draws them once and caches the result,
so the only operation that has to run on every vsync is a cheap linear
interpolation, plus CMS/dithering.
On my machine, this is a huge speedup for 24 Hz content (on a 60 Hz
monitor), up to 120% faster. (The speedup is not quite 250% because of
the overhead that the larger FBOs and CMS provides)
In terms of the implementation, this commit basically swaps
interpolation and upscaling - upscaling is moved to inter_program, and
interpolation is moved to the final_program.
Furthermore, the main bulk of the frame rendering logic (upscaling etc.)
was moved to a separete function, which is called from
gl_video_interpolate_frame only if it's actually necessarily, and
skipped otherwise.
Hopefully, this will really clear up how the thing is supposed to work
(and that it's not SVP, nor MVTools).
I also removed instances of the word "interpolation", since that's a
term that's easily misleading.
Finally, I expanded on smoothmotion-threshold since the purpose/meaning
was a bit confusing.
Update options like --sub-text-margin-y at runtime. These are somewhat
of a problem, because ass_set_selective_style_override() (intentionally)
does not override them. This should be fixed in libass (by providing
additional override modes), but for now this will do.
Also change the signature of mp_ass_configure(), so we can get access to
the track.
Additionally, drop the redundant setting of the style Alignment (it's
overwritten by mp_ass_set_style()).
See #1622 (again).
Overwrite the alignment applied by the OSD style. Additionally, remove
the initialization of the Alignment field in create_ass_track(); the
value is always overwritten by mp_ass_set_style() later.
Fixes#1626.
There was some logic to set certain things on init only. Not sure why
this was done (saving some cheap calculations?) - but since the next
call would override these style settings by applying the usual subtitle
style, I don't think this was intended.
GLES2 randomly does not support the transpose parameter in matrix
uniform calls. So we have to do this manually. Sure it was worth to
mutilate the standard just so all these shitty SoC vendors can safe 3
lines of code.
(Obviously trying to handle all of GLES2 to GL 4.x in a single codebase
was a mistake.)
GLES2 shaders do not have line continuation characters. Abuse the
HAVE_ARRAYS define to exclude code which uses arrays, and which also
happens to cover all code that defines multi-line macros. (So yes, this
is a hack.)
Remove coded_width and coded_height. This was originally added in commit
fd7dde40, when BITMAPINFOHEADER was killed. The separate fields became
redundant in commit e68f4be1. Remove them (nothing passed to the
decoders actually changes with _this_ commit).
Basically abuse the style override mechanism meant for ASS
(mp_ass_set_style()) to update text subtitle styling at runtime too.
This even has the advantage that the style will be overridden, even if
the text subtitle converted (like sd_lavc_conv.c) dares to add a fixed
style in the styles section.
Probably helps with #1622.
This is essentially what it is, and it's a useful for windowing or
downscaling. For upscaling we already have bilinear, no need to cause
extra confusion between biliner and bilinear_slow.
Also made it a bit more well-behaved.
These are EWA-based versions of the keys B/C splines, of which mitchell
is already a member. They are slightly softer and slightly sharper than
mitchell, respectively.
Very easy to define in terms of things we already have.
mitchell, hermite and catmull_rom are all B/C splines and can share the
code which was already written for mitchell. This just redefines them in
terms of that.
This adds a small check for candidates that could potentially be inside
the radius, but aren't necessarily. This speeds up performance by a
negligible amount on my hardware, but it's mainly a prerequisite for a
further change (using a larger true radius).
This echanges the two events hForceFeed/hFeedDone for hResume. This
like the last commit makes things more deterministic.
Importantly, the forcefeed is only done if there is not already a full
buffer yet to be played by the device. This should fix some of the
problems with exclusive mode.
This commit also removes the necessity to have a proxy to the
AudioClient object in the main thread.
fixes#1529
This makes things a bit more deterministic. It ensures that the audio
thread isn't doing anything between IAudioClient_Stop(),
IAudioClient_Reset() and setting the sample_count to 0.
Buffer overfilling on resume is still a problem in exclusive mode (see
next commit).
Previously, this was based on some arbitrary range 1-100, cut off for
no particular reason, and also defined in such a way that higher values
= *less* smoothness. Since it wasn't multiplied by e in the code, the
default had to be 10*e = 28.8539...
Now, it's sane: 1.0 = default, higher = blurrier.
This filter isn't supposed to have a second parameter in the first
place, all literature only uses a single parameter alpha in both places.
The second parameter doesn't even do anything other than adding a
constant factor, which is normalized by the LUT calculation either way.
This is done mainly for consistency, since all of the EWA filters share
similar properties and it's important to distinguish them for
documentation purposes.
No point in duplicating this check all over the place. No point in
really having it in the first place, to be perfectly honest, j1 should
not be THAT badly behaved.
A while ago, we made mpv output the native number of channels by
default, instead of stereo. I assumed this was not wanted for encode
mode.
This commit keeps the assumption, but allows setting the number of audio
output channels at all, instead of always forcing stereo. (Pretty much a
hack.)
The original filter window was design for a radius based on the true
zero, but we always cut it off at our selection of radius either way (by
necessity, due to the square matrix we sample from).
This window is tweaked from the original (true radius) to our actual
cut-off radius, and hence improves the result in a few edge cases. The
main win is the reduction of code complexity, since we no longer need to
know what the true radius actually is.
Commit f54220d9 attempted to improve this, but it got worse. Now there
was a crash when ytdl_hook.lua added external tracks. This happened
because close_unused_demuxers() assumed that sources[0] was the main
demuxer (so that it didn't close it). This assumption failed, because
the ytdl script can add external tracks before the main file is loaded.
The easy fix would have been to check for master_demuxer, and not i==0.
But instead give up on the old idea, make some stricter assumptions how
demuxers and external tracks map, and simplify the code.
Do timeline building (scanning & opening reference files for ordered
chapters, and more) in a thread. As a result, this process can actually
be stopped without having to kill the player.
This is pretty simple: just reuse the demuxer opening thread. We have
to give up on the idea that open_demux_reentrant() is reusable, though.
(Althoughthe timeline readers still need some fixes before they react to
the quit request.)
These functions do blocking work on a separate thread, but wait until
they return. So they are not async or non-blocking. But they do react to
user-input and client API accesses, which makes them reentrant.
Check the scanf() return value, and don't continue if it doesn't find
both numbers (can happen with GLES 1.0). Also, some implementations can
return NULL from glGetString() if something is "broken".
This is a variation of ewa_lanczos that is sinc-windowed instead of
jinc-windowed. Results are pretty similar, but the logic is simpler.
This could potentially replace the ugly ewa_lanczos code.
It's hard to tell, but from comparing stills I think this one has
slightly less ringing than regular ewa_lanczos.
I've reworked pretty much all the logic to correspond to what the theory
actually describes. With this commit, playback is wonderfully smooth on
my machine.
Some of the hacks were not applied if the file format was forced. Commit
37a0c914 moved them to a table, which is checked with normal probing
only.
Fixes#1612 (DVD forces mpeg, which in turn has to export native stream
IDs specifically).
Do some code restructuring on the way. For example, the probescore can
simply be set to the correct initial value, instead of checking whether
it was set at all.
Starting with waf 1.8.6 (in Python 3), the hcode variable isn't a
string, but a byte string.
This commit adds the solution proposed in the upstream waf bug report:
https://code.google.com/p/waf/issues/detail?id=1535
It seems a bit overly verbose, but on the other hand, this solution has
the chance of being most correct/compatible.
Fixes#1604.
Nobody should use an older version. It's perfectly backwards and forward
compatible, so distros have no excuse not to package a recent version.
Older versions lack tons of bug fixes (some of them crashing bugs, and
potentially security relevant).
With love to Debian, which is still on 0.10.2.
Whatever the hell that is. FFmpeg tries to open any files with .bin file
extension with this demuxer (unless it finds a better demuxer), and then
reads the whole damn file, along with spamming dumb crap.
This message will be printed relatively often once EOF is reached. In
some cases this is rather annoying, for example when playing HLS. (With
HLS, the stream is just a playlist file, while libavformat opens actual
media files without mpv's knowledge, so the cache is completely useless
and hits EOF instantly.)
That it retries reading is apparently a good thing: at least local files
can grow, and even after the player got the EOF, playback _could_ be
resumed by basically polling and detecting that there is more data. So
I'm not changing this behavior yet.
Includes some logic for not starting the demuxer thread for fully read
subtitles. (Well, the cache will still waste _lots_ of resources, and
the cache always has to be created, because we don't know whether it'll
be needed _before_ opening the file.)
See #1597.
An attempt to make format-specifics more declarative. (In my opinion,
all of this should be either provided by libavformat, or should not be
needed.)
I'm still leaving many checks with matches_avinputformat_name(), because
they're so specific.
Also useful for the following commit.
Treat an empty string as unset. The fact that the option values can be
NULL is merely weirdness due to how the option parser works (it
unfortunately doesn't initialize string fields to non-NULL).
Add MP_KEY_MOUSE_ENTER to the ignored input if the user has disabled
mouse input. Remove one instance of code duplication, and add a
MP_KEY_IS_MOUSE_MOVE macro to summarize events that are caused by moving
the mouse.
The Matroska timeline code was the only thing which still used the
demuxer.type field. This field explicitly identifies a demuxer
implementation. The purpose of the Matroska timeline code was to reject
files that are not Matroska. But it already forces the Matroska format,
meaning loading will explicitly only use the Matroska demuxer. If the
demuxer can't open the file, no other demuxer will be tried, and thus
checking the field is redundant.
The change in demux_mkv_timeline.c removes the if condition, and
unindents the if body.
Only demux_cue and demux_edl used it. It's a weird field and doesn't
help with anything anymore - by now, it only saves a priv context in the
mentioned demuxers. Reducing the number of confusing things the demuxer
struct has is more important than minimizing the code.
Move the implementation, of which most was in tl_cue.c, to demux_cue.c.
Currently, this is illogical, because tl_cue.c still accesses MPContext.
This is going to change, and then it will be better if everything is in
demux_cue.c. This is only a separate commit to distinguish code movement
and actual work; the next commit will do the actual work.
Instead of accessing MPContext in player/timeline/*, create a separate
context struct, which the timeline loaders fill out. It turns out that
there's not much in the way too big MPContext that these need to access.
One major PITA is managing (and closing) the set of open demuxers. The
problem is that we need a list of all demuxers to make sure no unneeded
streams are enabled.
This adds a callback to the demuxer_desc struct, with the intention of
leaving to to the demuxer to call the right loader, instead of
explicitly checking the demuxer type and dispatching manually in common
code. I also considered making the timeline part of the demuxer state,
but decided against: it's too much of a mess wrt. memory management and
threading, and also doesn't make it clear who owns the child demuxers.
With the struct timeline decoupled from the demuxer state, it's at least
somewhat clear that the child demuxers are independent from the "main"
demuxer.
The actual changes to player/timeline/* are separated in the following
commits, because they're quite verbose. Some artifacts will be removed
later as soon as there's only 1 timeline loading mechanism.
Also effects some other cases.
The real reason for this is for keeping track of which demuxers can be
closed (see following commit). Since I don't want to use reference
counting for this, some sort of simplistic mark-and-sweep is done to
determine whether a demuxer is still needed.
The HLs protocol consists of a "playlist" main file, which mpv downloads
and passes to the HLS demuxer. The HLS demuxer actually requests segment
files containing media data on its own. The packets read from the
demuxer have a source file position set, but it's not from the main
file. This leads to a strange effect: as a last fallback, the player
will calculate the approximate playback position from the file
position/size ratio, and since the main file is tiny, this will always
show 100%. Fix this by resetting the packet file position.
This doesn't affect the case when HLS actually reports a duration.
Now --ass-use-margins doesn't apply to normal subtitles anymore. This is
probably the inverse from the mpv behavior users expected so far, and
thus a breaking change, so rename the option, that the user at least has
a chance to lookup the option and decide whether the new behavior is
wanted or not.
The basic idea here is:
- plain text subtitles should have a certain useful defalt behavior,
like actually using margins
- ASS subtitles should never be broken by default
- ASS subtitles should look and behave like plaintext subtitles if
the --ass-style-override=force option is used
This also subtly changes --sub-scale-with-window and adds the --ass-
scale-with-window option. Since this one isn't so important, don't
bother with compatibility.
You can set in which "corner" the OSD and subtitles are shown. I'd
prefer it a bit more general (so you could set the alignment using
a factor), but the libass API does not provide this.
Until now, they used exactly the same defaults for the styling options.
The defaults were shared, so it was impossible to have different
defaults. Change this. This requires duplicating the full default
struct, even for settings that are the same. The list of options is
still shared, though.
The previous commit effectively fixes the mess caused by 'config' vs.
'mpv.conf', and the hack introduced by commit e01a6dac and extended by
commit db167cd4 isn't needed anymore.
Actually, it's pretty simple to look for multiple filenames at once,
since mp_find_all_config_files() is already a bit "special" anyway.
See #1569. Reverts most of commit db167cd4 (keeps osx-bundle.conf).
A recent behavior change in libavcodec's h264 decoder keeps at least 1
surface even after avcodec_flush_buffers() has been called. We used to
flush the decoder in order to make sure all surfaces are free'd, so that
the hw decoder can be safely uninitialized. This doesn't work anymore.
Fix it by closing the AVCodecContext before the hw decoder is
uninitialized. This is actually simpler and more robust. It seems to be
well-supported too.
Fixes invalid read accesses with vaapi-copy and dxva2-copy. These
destroyed the hwdec API fully on uninit, and could not deal with
surfaces surviving the decoder.
Probably fixes#1587.
The way the AO wakes up the playloop has nothing to do with events;
instead we must query the events on the AO once the playloop was woken
up. Querying the events in every playloop iteration is thus the correct
way to do this.
If the previous subtitle packet is too far back, and the refresh seek
won't pick it up, and the packet never comes again. As a consequence,
the refresh mode was never stopped on the subtitle stream, which caused
all packets to be discarded.
Fix by assuming the file position is monotonically increasing; then it
will resume even if a packet _after_ the intended resume point is
returned. This introduces a new requirement on how the demuxer behaves.
(I'm not sure if mp4 actually satisfies this requirement in all cases.)
Fixes a regression introduced by commit f9f2e1cc.
This commit adds notifications for hot plugging of devices. It also extends
the old behaviour of the `audio-out-detected-device` property which is now
backed by the hotplugging code. This allows clients to be notified when the
actual audio output device changes.
Maybe hotplugging should be supported for ao_coreaudio_exclusive too, but it's
device selection code is a bit fragile.
Apparently there's at least one distro which ships a /etc/mpv/mpv.conf
file (mpv doesn't install such a file). This breaks config files named
'config' located in the user's mpv config directory, because mpv first
loads files named 'config' and then 'mpv.conf'. There is no mechanism
for putting files with different names into the same config path order.
(Even worse, that mpv.conf file only set an option to the default value.
Why do distros always do very stupid things?)
Print a warning on collisions.
Although using 'config' was well-supported, supporting both names is
starting to become messy, so deprecate 'config' and print a warning if
one is found.
At least we will be able to remove the whole mess once 'config' files
are ignored...
This also affects the osx-bundle, which intentionally used these not-so-
optimal semantics. Solve it in a different way. (Unfortunately with an
ifdef - it's not required, but having to explain everyone why mpv tries
to load a osx-bundle.mpv file on Linux and Windows would consume
energy.)
Closes#1569.
Makes all keys documented in XF86keysym.h mappable. This requires the
user to deal with numeric keycodes; no names are queried or exported.
This is an easy way to avoid adding all the hundreds of XF86 keys to
our X11 lookup table and mpv's keycode/name list.
Happens to fix#1581 due to an unfortunate interaction with the way the
VO does not react to commands for a while if a video frame is queued.
Slightly improves other situations as well, if the client spams mpv with
commands during playback.
This removes the delay when switching audio tracks in mkv or mp4 files.
Other formats are not enabled, because it's not clear whether the
demuxers fulfill the requirements listed in demux.h. (Many formats
definitely do not with libavformat.)
Background:
The demuxer packet cache buffers a certain amount of packets. This
includes only packets from selected streams. We discard packets from
other streams for various reasons. This introduces a problem: switching
to a different audio track introduces a delay. The delay is as big as
the demuxer packet cache buffer, because while the file was read ahead
to fill the packet buffer, the process of reading packets also discarded
all packets from the previously not selected audio stream. Once the
remaining packet buffer has been played, new audio packets are available
and you hear audio again.
We could probably just not discard packets from unselected streams. But
this would require additional memory and CPU resources, and also it's
hard to tell when packets from unused streams should be discarded (we
don't want to keep them forever; it'd be a memory leak).
We could also issue a player hr-seek to the current playback position,
which would solve the problem in 1 line of code or so. But this can be
rather slow.
So what we do in this commit instead is: we just seek back to the
position where our current packet buffer starts, and start demuxing from
this position again. This way we can get the "past" packets for the
newly selected stream. For streams which were already selected the
packets are simply discarded until the previous position is reached
again.
That latter part is the hard part. We really want to skip packets
exactly until the position where we left off previously, or we will skip
packets or feed packets to the decoder twice. If we assume that the
demuxer is deterministic (returns exactly the same packets after a seek
to a previous position), then we can try to check whether it's the same
packet as the one at the end of the packet buffer. If it is, we know
that the packet after it is where we left off last time.
Unfortunately, this is not very robust, and maybe it can't be made
robust. Currently we use the demux_packet.pos field as unique packet
ID - which works fine in some scenarios, but will break in arbitrary
ways if the basic requirement to the demuxer (as listed in the demux.h
additions) are broken. Thus, this is enabled only for the internal mkv
demuxer and the libavformat mp4 demuxer.
(libavformat mkv does not work, because the packet positions are not
unique. Probably could be fixed upstream, but it's not clear whether
it's a bug or a feature.)
Until now, some packets could return the same file position if they were
split off from a Matroska-level packet. This was perfectly fine, because
the file position isn't used for anything overly important (it uses it
to estimate playback position if no other information is available). The
following commit will use the demux_packet.pos field as unique ID (as a
simplification), so make the demuxer export more finegrained
information.
Also, the last_filepos field didn't have to be global, at least not
anymore.
Commit e920a00eb assumed that terminate_cocoa_application() actually
would exit. But apparently that is not always the case; e.g. mpv --help
will just hang. The old code had a dummy exit(0), which was apparently
actually called. Fix by explicitly exiting if mpv_main() returns and
terminate_cocoa_application() does nothing.
Requested. See manpage additions.
This also makes the magical loop_times constants slightly saner, but
shouldn't change the semantics of any existing --loop option values.
Setting the input context is always called, both in cplayer and libmpv,
and under HAVE_COCOA. Unsetting the input context was done only the
cplayer uninit call. Also it was under HAVE_COCOA_APPLICATION, so it was
not unset in libmpv (dangling pointer).
The code in main.c calls exit() explicitly, but the code is actually
easier to follow by simply exiting from main() instead. The exit() call
in av_log.c happens only on severely broken builds, so replace it with
abort().
(Shuts up rpmlint warnings.)
This requires jumping through multiple hoops on fire. Since the
PulseAudio API is virtually undocumented, I'm not sure if this is
correct either. We only react to sink events, and only to the NEW/REMOVE
events. CHANGE events are ignored, because PulseAudio fires them far too
often - even if the system is completely idle! If pa_sink_info.name can
change, we're in trouble. pa_sink_info.description is not so important,
but it'd also be a bit un-nice if it can change, and we don't update it.
The weird way how the actual AO and the hotplug context share the same
struct (ao) comes in handy here, although context_success_cb() still had
to be duplicated from success_cb() - the unused argument has a different
type.
Not very important for the command line player; but GUI applications
will want to know about this.
This only adds the internal API; support for specific audio outputs
comes later.
This reuses the ao struct as context for the hotplug event listener,
similar to how the "old" device listing API did. This is probably a bit
unclean and confusing. One argument got reusing it is that otherwise
rewriting parts of ao_pulse would be required (because the PulseAudio
API requires so damn much boilerplate). Another is that --ao-defaults is
applied to the hotplug dummy ao struct, which automatically applies such
defaults even to the hotplug context.
Notification works through the property observation mechanism in the
client API. The notification chain is a bit complicated: the AO notifies
the player, which in turn notifies the clients, which in turn will
actually retrieve the device list. (It still has the advantage that it's
slightly cleaner, since the AO stuff doesn't need to know about client
API issues.)
The weird handling of atomic flags in ao.c is because we still don't
require real atomics from the compiler. Otherwise we'd just use atomic
bitwise operations.
This is a small oversight. The client name (as set on command line
options or, more importantly, the client API) was not set when listing
devices e.g. via the "audio-device-list" property.
Might or might not fix#1578.
Also adjust the log level for an unrelated message.
This reverts commit acc5e8f574.
As expected, some didn't like this. Others won't like this revert.
Whatever.
See #1561.
This should go into mpv 0.8.0 before it's released.
In my opinion the artifacts created by af_scaletempo on extreme slowdown
(50% or so) are too bothersome - but users disagree. So use
af_scaletempo on any speed changes, not just on speedup.
Filters which merely wrap libavfilter (for user-compatibility) like
vf_gradfun had a "lavfi-enable" suboption, which could disable
libavfilter usage. Since none of these filters has an internal
implementation anymore, this was completely useless.
rubberband_available() can return a negative value, which we assigned to
a size_t variable, leading to the frame allocation to fail. This could
spam "Error filtering frame.". (That it spams this instead of exiting
should probably also be considered a bug.)
At least in the realtime mode and in our case, a negative return value
should not have any different meaning from a 0 return value, in
particular because we call rubberband_get_samples_required() or set the
"final" parameter for rubberband_process() to continue/stop processing.
After some testing, I am fairly convinced that these defaults sound
better than the previous settings. This also eliminates some issue
with random crackling and noise.
Also remove the `stretch` option since it has no effect in
realtime mode.
The previous commit on this filter accidentally removed the
RubberBandOptionProcessRealTime option. Without it, the lib prints a
warning and passes the audio through.
Also add the RubberBandOptionSmoothingOn option back. Though for some
reason the output sounds still very wrong.
Remove the confusing crap that allowed a filter using the libavfilter
bridge to be compiled without libavfilter. Instead, compile the wrappers
only if libavfilter is enabled at compile time.
The only filter which still requires it is vf_stereo3d (unfortunately).
Special-case this one. (The whole filter and how it interacts with lavfi
is pure braindeath anyway.)
librubberband exports a big load of options. Normally, the default
settings (whether they're librubberband defaults or our defaults) should
be sufficient, but since I'm not so sure about this, making it
configurable allows others to figure it out for me.
The problem here is that librubberband can buffer an arbitrary amount
of data, but at the same time doesn't provide a way to query how much
data is buffered. So we keep track of this manually, assuming that
librubberband tries to reach the requested time ratio for input and
output (which is probably true).
The disadvantage is that rounding errors could accumulate over time.
(Maybe it should try to round towards keeping the time ratio.)
In theory it could happen that draining on EOF happens incrementally,
and then the unconditional reset could have dropped the remaining
buffered audio.
If "--af=rubberband" is used, librubberband will be used to speed up or
slow down audio with pitch correction.
This still has some problems: the audio delay is not calculated
correctly, so the audio position jitters around by a few milliseconds.
This will probably ruin video timing.
This avoids potentially dropping some small amount of audio data
buffered in filters.
Reinit can be skipped only if the filter is af_scaletempo (which maps to
AF_CONTROL_SET_PLAYBACK_SPEED). The other case using af_lavrresample is
much more complicated due to filter chain politics.
Also, changing speed between 1.0 and something higher typically inserts
or removes the filter, so this obviously requires reinitialization. It
can be prevented by forcing the filter with --af=scaletempo.
Staring at the code a bit, it turns out that changing speed without
losing state is quite easy. The initialization code is big and
complicated, but most of it is specific only to the configured audio
format, not the speed.
Refactor the code so that changing speed at runtime could work. (It's
not actually used yet - the player code still does a complete reinit.
This will be fixed in the next commit.)
The "if (s->speed_tempo == s->speed_pitch)" looks a bit strange, but
does the same thing as the code did before: speed can be changed only if
exactly one flag is set. If both are set or none, speed can't be
changed.
This code skipped initialization if no speed/pitch change was to be
applied.
It also didn't force conversion of the audio to a supported format,
which is probably the most important case in context of compatibility.
With this change applied, af_scaletempo will always force format
conversion.
To make the change less disruptive, make the filter detach if
unconvertable formats are used. Some users use spdif and also have
"af=scaletempo" in their config, so better not completely break this.
In the case the filter was added with the "speed=both" suboption, the
filter also detached itself in this case; but it's an obscure case, so I
don't care about that.
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Diogo Franco (Kovensky)