In the past, --video-unscaled also disabled zooming and aspect ratio
corrections. But this didn't make much sense in terms of being a useful
option. The new behavior just sets the initial video size to be
unscaled, but it's still affected by zoom commands and aspect ratio
corrections.
To get the old behavior back, --video-aspect=0 --video-zoom=0 need to be
added as well (in the general case). Most of the time it should not make
a difference though.
Also, there seems to have been some additional dst_rect clamping code
inside src_dst_split_scaling that didn't seem to either be necessary nor
ever get triggered. (The code immediately above it already makes sure to
crop the video if it's larger than the dst_rect)
No idea why it was there, but I just removed it.
Apply basic transformations like rotation by 90° and mirroring when
sampling from the source textures. The original idea was making this
part of img_tex.transform, but this didn't work: lots of code plays
tricks on the transform, so manipulating it is not necessarily
transparent, especially when width/height are switched. So add a new
pre_transform field, which is strictly applied before the normal
transform.
This fixes most glitches involved with rotating the image.
Cropping and rotation are now weirdly separated, even though they could
be done in the same step. I think this is not much of a problem, and
has the advantage that changing panscan does not trigger FBO
reallocations (I think...).
Typically happens with some implementations if no context is currrent,
or is otherwise broken. This is particularly relevant to the opengl_cb
API, because the API user will have no other indication what went wrong.
The underlying intention of this code is to make changing
--videotoolbox-format at runtime work. For this reason, the format can't
just be statically setup, but must be read from the option at runtime.
This means the format is not fixed anymore, and we have to make sure the
renderer is property reinitialized if the format changes. There is
currently no way to trigger reinit on this level, which is why the
mp_image_params.hw_subfmt field was introduced.
One sketchy thing remains: normally, the renderer is supposed to be
involved with VO format negotiation, which would ensure that the VO
can take the format at all. Since the hw_subfmt is not part of this
format negotiation, it's implied the get_vt_fmt() callback only
returns formats supported by the renderer. This is not necessarily
clear because vo_opengl checks this with converted_imgfmt separately.
None of this matters in practice though, because we know all formats
are always supported.
(This still requires somehow triggering decoder reinit to make the
change effective.)
For hwaccel formats, mp_image will merely point to a hardware surface
handle. In these cases, the mp_image_params.imgfmt field describes the
format insufficiently, because it mostly only describes the type of the
hardware format, not its underlying format.
Introduce hw_subfmt to describe the underlying format. It makes sense to
use it with most hwaccels, though for now it will be used with the
following commit only.
Until now, the presence of the process_image() callback was used to set
a delay queue with a hardcoded size. Change this to a vd_lavc_hwdec
field instead, so the decoder can explicitly set this if it's really
needed.
Do this so process_image() can be used in the VideoToolbox glue code for
something entirely unrelated.
Some functions which expected a codec name (i.e. the name of the video
format itself) were passed a decoder name. Most "native" libavcodec
decoders have the same name as the codec, so this was never an issue.
This should mean that e.g. using "--vd=lavc:h264_mmal --hwdec=mmal"
should now actually enable native surface mode (instead of doing copy-
back).
The sync-by-display mode relies on using the vsync statistics for
timing. As a consequence discontinuities must be handled somehow. Until
now we have done this by completely resetting these statistics.
This can be somewhat annoying, especially if the GL driver's vsync
timing is not ideal. So after e.g. a seek it could take a second until
it locked back to the proper values.
Change it not to reset all statistics. Some state obviously has to be
reset, because it's a discontinuity. To make it worse, the driver's
vsync behavior will also change on such discontinuities. To compensate,
we discard the timings for the first 2 vsyncs after each discontinuity
(via num_successive_vsyncs). This is probably not fully ideal, and
num_total_vsync_samples handling in particular becomes a bit
questionable.
Tuning it in a way to be actually useful is too much effort.
As alternative, there's the "buffering" detection, which operates on a
much higher level. The only disadvantage is that it's harder to guess
for the user whether this is a network problem, or if e.g. libavformat
is probing too much data when opening a stream. Maybe the cache-speed
property is helpful here.
For now, do not remove the associated code, but just silence the
warning.
Fixes#3019.
It's pretty "unfriendly" and causes too many issues. (Probably. At least
they're more obvious to a user than e.g. broken frame timing.)
Potentially we could apply heuristics like applying this only on
fullscreen, but let's not. It's up to the user to configure this to
get best results.
Fixes#2997.
The past behavior was a bit weird, especially when zooming out. There
was no simple way to zoom in or out in consistent increments using
keybindings alone.
The new behavior preserves most of the old behavior's semantics but
scales out to infinity better. It coincidentally also makes it
really easy to get clean power of 2 ratios (e.g. 2x, 4x, 8x and their
inverses).
Fixes#3004.
I got a report that the build on a recent aarch64 Linux kernel failed.
DVB support was detected, but errored on compilation:
In file included from ../stream/stream_dvb.c:57:0:
../stream/dvbin.h:72:5: error: unknown type name 'fe_bandwidth_t'
fe_bandwidth_t bw;
Make the test stricter, which should take care of this. (I couldn't find
out what exactly triggered the failure, nor could I attempt to reproduce
it.)
The change in stream/dvbin.h is to make sure that this isn't caused by
incorrect header inclusion. It now includes the same files as the
configure test.
There is an obscure feature which requires essentially reordering PTS
from different packets.
Unfortunately, libavcodec introduced a ridiculously shitty API for
this, which works very much unlike the audio/video API. Instead of
simply passing through the PTS, it wants to fuck with it for no reason,
and even worse, fucks with other fields and changes their semantivcs
(??????). This affects AVSubtitle.end_display_time. This probably will
cause issues for us, and I have no desire to find out whether it will.
Since only PGS requires this, and it happens not to use
end_display_time, do it for PGS only.
Fixes#3016.
Basically, this information is useless, because some muxers (hurr
libavformat) write bogus information anyway. This means if we e.g. see
PGS packets in mkv with duration explicitly set to 0, we must not trust
that value anyway. (The FFmpeg API problem is leaking into files, how
nice.)
This makes the black point closer (chromatically) to the white point, by
ensuring channels keep their consistent brightness ratios as they go
down to zero.
I also raised the 3DLUT version as this changes semantics and is a
separate commit from the previous one.
This commit refactors the 3DLUT loading mechanism to build the 3DLUT
against the original source characteristics of the file. This allows us,
among other things, to use a real BT.1886 profile for the source. This
also allows us to actually use perceptual mappings. Finally, this
reduces errors on standard gamut displays (where the previous 3DLUT
target of BT.2020 was unreasonably wide).
This also improves the overall accuracy of the 3DLUT due to eliminating
rounding errors where possible, and allows for more accurate use of
LUT-based ICC profiles.
The current code is somewhat more ugly than necessary, because the idea
was to implement this commit in a working state first, and then maybe
refactor the profile loading mechanism in a later commit.
Fixes#2815.
FFmpeg partially merged the API change. It added the AVCodecParameters
definition, but not the AVCodecContext.codecpar field. The new code
compiles only with the API fully merged, so adjust the check.
Encoding mode uses deprecated API. See previous commit. Encoding mode
will stop working/compiling at some point in the future, so unless
someone fixes the encoding code, it will stay disabled by default.
(Note that the deprecations are not merged in FFmpeg yet, but they will
soon. They've been deprecated in Libav for a while now.)
AVFormatContext.codec is deprecated now, and you're supposed to use
AVFormatContext.codecpar instead.
Handle this for all of the normal playback code.
Encoding mode isn't touched.
This was changed in 2014, so I suppose users will usually have a FFmpeg
release which includes the corresponding upstream change. If not, well
too bad for those MicroDVD-obsessed users.
Also don't try to retrieve the default framerate as exported by the
demuxer, and instead hardcode it and trust it won't ever change. this
avoids that we have to deal with a larger mess in the codecpar commit.
I don't trust it one bit, and it's a bother with the codecpar change.
If it turns out to be important for some file formats, it could be
added back (or FFmpeg fixed).
This commit adds the d3d11va-copy hwdec mode using the ffmpeg d3d11va
api. Functions in common with dxva2 are handled in a separate decode/d3d.c
file. A future commit will rewrite decode/dxva2.c to share this code.
Commit 57506b27 accidentally broke this. The status (including the
usually always active demuxer cache) should be shown only if the stream
cache is actually enabled.
This reverts commit 503c6f7fd6.
There are situations where some decoders (MF apparently) always require
a timestamp. Also, this makes bitrate estimation more granular than
necessary. It seems it's better to try to detect fiels with broken
default durations explicitly instead. Or maybe something should be
added to smooth audio timestamps after filters.
This also draws it after color management etc. In a nutshell, this
change makes the transparency checkerboard independent of upscaling,
panning, cropping etc. It will always be the same apparent size and
position (relative to the window).
It will also be independent of the video colorspace and such things.
(Note: This might cause white imbalance issues if playing a file with a
white point that does not match the display, in absolute colorimetric
mode. But that's uncommon, especially in conjunction with transparent
image files, so it's not a primary concern here)
Until now, we've let the windowing backend decide. But since they
usually require premultiplied alpha, and premultiplied alpha is easier
to handle, hardcode it.
The recent changes fixed rotation handling, but reversed the rotation
direction. The direction is expected to be counter-clockwise, because
demuxers export video rotation metadata as such.
Don't assume EOF if we didn't try to read anything in the first place.
Fixes regressions in particular with low cache sizes, which triggered
the other code paths more often.
Instead of having a separate for each, which also requires separate
additional caching in the demuxer. (The demuxer adds an indirection,
since STREAM_CTRLs are not thread-safe.)
Since this includes the cache speed, this should fix#3003.
This would get stuck in reconfiguring the filter chain forever, because
params was mutated ("params.rotate = 0;"). This was used as input for
vf_reconfig(), but the filter chain input must always be equivalent to
the decoder output, or filter chain reconfiguration will be triggered.
The line of code to reset the rotation is from a time when this used to
work differently.
Also remove the unnecessary try_filter() parameter.
This has been completely broken since commit 93546f0c. But even before,
rotation handling did not make too much sense. In particular, it rotated
the contents of the cropped image, instead of adjusting the crop
rectangle as well. The result was that things like panscan or zooming
did not behave as expected with rotation applied.
The same is true for vertical flipping. Flipping is triggered by
negative image stride. OpenGL does not support flipping the image on
upload, so it's done as part of the rendering. It can be triggered with
--vf=flip, but other filters and even decoders could setup negative
stride to flip the image.
Fix these issues by applying transforms to texture coordinates properly,
and by making rotation and flipping part of these transforms.
This still doesn't work properly for separated scaling. The issue is
that we'd have to adjust how the passes are done. For now, pick a very
stupid solution by rotating the image to a FBO, and then scaling from
that. This has the avantage that the scale logic doesn't have to be
complicated for such a rare case. It could be improved later.
Prescaling is apparently still broken. I don't know if chroma
positioning works properly either. None of this should affect the case
with no rotation.
gl_transform_vec() assumed column-major, while everything else seemed to
assumed row-major memory organization for gl_transform.m. Also,
gl_transform_trans() seems to contain additional confusion.
This didn't matter until now, as everything has been orthogonal, this
the swapped matrix entries were always 0.
If the texture count is lower than 4, entries in va.textcoord[] will
remain uninitialized. While this is unlikely to be a problem (since
these values are unused on the shader side too), it's not nice and might
explain some things which have shown up in valgrind.
Fix by always initializing the whole thing.
E.g. "mouse 100 100 1 double" did not actually process the double-click,
because double-click emulation is on by default. So the user would have
to send two successive clicks instead. This is probably not expected, so
disable this weird logic for artificial input.
Fixes#2899.
In particular, this prevents subtitle packets from building up in the
subtitle queue if e.g. --vo=null is used. In this situation,
sub_get_bitmaps() is never called, and thus the segment never switched.
This also seems to help with flickering at segment switch boundaries (if
subs are supposed to be visible at the transition points).
In theory, this could trigger a switch too early, but the way VO and
subtitle renderer interact wrt. timing is a bit iffy anyway.
SEEK_HR is interpreted by demux_mkv.c, and enables subtitle preroll by
prefetching additional subtitle pakcets which might overlap with the
seek destination. This should make the case work when segment boundaries
fall into the middle of subtitle events.
This still usually leaves a flicker of at least 1 frame on start,
because dec_sub.c does not ensure that enough subtitles are read before
rendering after a segment switch. (Probably a WONTFIX.)
The default security descriptor for named pipes in Windows allows the
pipe to be opened for read access by the Everyone group and Anonymous
account, as well as low-integrity processes (like web browser renderer
processes.) This does not allow commands to be ran, but it does allow
events to be received.
I don't think any sensitive data is exposed by events, but that may not
always be the case and Lua plugins might change this, since they can
broadcast their own events with script-message. To be safe, this commit
sets a custom security descriptor on the named pipe which only allows
access from processes running under the same user account with an
integrity level greater than or equal to the one used by mpv.
Instead of reallocating almost all of the shader string several times
per pass, build it into a fixed buffer that will be reallocated as
needed.
While this still uses a linear search and full comparison of the shader
text, this will compare the shader's string length first before doing a
full comparison as a nice side effect. (That's also why the fragment
shader is compared first - it's more likely to be different for
different cache entries than the vertex shader stub.)
The mp_set_av_packet()/mp_pts_from_av() functions check whether the
timebase is set at all (i.e. AVRational.num!=0), so there's no need to
fiddle with pointers.
Until now, bstr_xappend_vasprintf() called vsnprintf() always twice:
once to determine how much output the call would produce, and a second
time to actually output the data to the (possibly resized) target
memory.
Change this so that it tries to output to the already allocated memory
first, and repeat the call only if allocation is required.
This is especially helpful, as bstr_xappend_vasprintf() is designed to
avoid reallocation when building strings. Usually, the second
vsnprintf() will happen only at the beginning, when the buffer hasn't
been extended to his largest needed size yet.
Not sure if there is a need to optimize this; but see the next commit.
JSON IPC works on Windows now, and although the transports for each
plaform have similar characteristics, they unfortunately have different
names (Unix domain sockets on Linux/Unix vs. named pipes on Windows.)
Hopefully this change better reflects the purpose of the option too,
since with --input-ipc-server, mpv acts as an IPC server that can
service many simultaneous clients (as opposed to --input-file, which can
only do one-to-one IPC.)
This implements the JSON IPC protocol with named pipes, which are
probably the closest Windows equivalent to Unix domain sockets in terms
of functionality. Like with Unix sockets, this will allow mpv to listen
for IPC connections and handle multiple IPC clients at once. A few cross
platform libraries and frameworks (Qt, node.js) use named pipes for IPC
on Windows and Unix sockets on Linux and Unix, so hopefully this will
ease the creation of portable JSON IPC clients.
Unlike the Unix implementation, this doesn't share code with
--input-file, meaning --input-file on Windows won't understand JSON
commands (yet.) Sharing code and removing the separate implementation in
pipe-win32.c is definitely a possible future improvement.
Glitches when resizing are still possible, but are reduced. Other VOs
could support this too, but don't need to do so.
(Totally avoiding glitches would be much more effort, and probably not
worth the trouble. How about you just watch the video the player is
playing, instead of spending your time resizing the window.)
Should reflect I/O speed.
This could go into the terminal status line. But I'm not sure how to put
it there, since it already uses too much space, so it's not there yet.
The old algorithm produced results which were not uniformly distributed,
i.e. some particular shuffles were preferred over others.
The new algorithm is an implementation of the Fisher-Yates shuffle which
is guaranteed to shuffle uniformly given a sufficiently uniform rand()
and ignoring potential floating-point errors.
Signed-off-by: wm4 <wm4@nowhere>
Until now, we have tried to create a GL 3.0 context. The main reason for
this is that many Mesa-based drivers did not support anything better.
But some drivers (Mesa AMD) will not report a higher OpenGL version,
because their compatibility mode is restricted. While later GL features
are reported as extensions just fine, there doesn't seem to be a way to
determine or enable higher GLSL versions.
Add some more shitty hacks to try to deal with this messed up situation,
and try to probe each interesting GL version separately (starting with
3.3, then 3.2 etc.). Other backends might suffer from similar problems,
but these will have to deal with it on their own.
Probably fixes#2938, or maybe not.
Prevents an infinite loop of configure event and set_fullscreen
request on Weston and other compositors respecting the protocol.
Fixes#2817
This reverts commit eb6b2b6e50.
This colorspace has been historically used as a calibration target for
most digital projectors and sees some involvement in the UltraHD
standards, so it's a useful addition to mpv.
This changes behavior somewhat. The old behavior can be restored by
running "mp.use_suspend=true". It was originally introduced for the OSC,
but I can't reproduce whatever misbehavior I was seeing.
(See mp.suspend()/resume() for explanations what the suspend mechanism
does.)
converted_imgfmt will be used by the renderer logic to build an
appropriate shader chain. It doesn't influence the format of any
textures. Thus it doesn't matter whether the hw video surface is mapped
as RGB or RGBA. What matters is if the video actually contains alpha or
not. Since virtually all hardware decoder do not support alpha in any
way, this can be hardcoded as "no alpha".
This avoids unnecessary GPU work.
This also gets rid of the kind of hard to read texture swizzle setup and
turns it into something dumber.
Assumes that we don't create any FBOs with 2 channel formats. (Only the
video source textures are handled by this commit.)
This is particularly useful for opus which allows only a fairly restrictive set
of samplerates. If the codec doesn't provide a list of samplerates, just
continue to try the requsted one and hope for the best.
fixes#2957
This function chooses the best match to a given samplerate from a provided
list. This can be used, for example, by the ao to decide what samplerate to use
for output.
* Use the update-core command
* Add --check-c-compiler=gcc to be safe
* Add warning about potential pitfalls of adding C:\msys2\mingw64\bin to %PATH%
* Recommend winpty
* Add note about ANGLE
Previously, gl->DXOpenDeviceNV was called twice using dxva2 with dxinterop. AMD
drivers refused to allow this. With this commit, context_dxinterop sets its own
implementation of MPGetNativeDisplay, which can return either a
IDirect3DDevice9Ex or a dxinterop_device_HANDLE depending on the "name" request
string. hwdec_dxva2gldx then requests both of these avoiding the need to call
gl->DXOpenDeviceNV a second time.
Drag&drop mechanisms typically support multiple types for the drop data.
Move most of the logic which types are accepted and preferred to
event.c, where the data is also interpreted.
(Maybe sorting the types by assigning scores is over-engineered, since
they're already sorted by preference, but it's actually not much more
code.)
Not very interesting/meaningful yet, but preparation for the next
commit.
Reduces VO access and makes the code more self-contained. (One day the
windowing backend code should not access the VO anymore. We're just not
quite there yet.)
Instead of displaying it only on playback start (or after switching
tracks), always display it even after a seek.
This helps with --lavfi-complex. You can now overlay e.g. audio
visualizations over cover art, and it won't break after a seek.
The downside is that this might make seeks with huge cover art slower.
There is also a glitch on seeking: since cover art pictures always
have timestamp 0, the playback time will be 0 for a moment after seek,
and then revert to audio PTS (as video is considered EOF). This is also
due to how lavfi's overlay filter behaves. (I'm not sure how to tell
lavfi that it's just a single frame.)
Like dxinterop, this uses StretchRect or RGB conversion. This is unavoidable as
long as we use the dxva2 API, as there is no way to access the raw hardware
decoded Direct3D9 surfaces.
This is simpler, because it doesn't have to wait from both threads for
synchronization.
Apart from being simpler/cleaner, this serves vague plans to stop/start
the demuxer thread itself automatically on demand (for the purpose of
reducing unneeded resource usage).
This pause stuff is bothersome and is needed only for a few corner-
cases. This commit removes it from the demuxer public API and replaces
it with a demux_run_on_thread() function and refactors the code which
needed demux_pause(). The next commit will change the implementation.
Changing the byte stream position without cooperation of the demuxer
seems a bit insane, and is certainly useless. A user should do factor
seeks instead. For formats like ts, this will actually translate to byte
seeks, while treating the rest of the playback chain a bit more
gracefully. With this argument, remove write access to this property.
If someone really complains, proper byte seeks could be added as seek
mode (although I'm going to need a convincing argument for this).
Read access changes too, but in a more subtle way.
No need to have them everywhere. The only exception/annoyance is
MAX_OSD_PARTS, which is now basically duplicated (and at runtime
initialization is checked with an assert()).
Until now, there was only 1 global ASS overlay that could be set by all
scripts. This was often perceived as bug when multiple scripts tried to
set their own ASS overlay.
This was kind of hard to solve because the script could set its own ASS
PlayResX/Y, which makes it impossible to share a single ASS_Renderer for
multiple scripts. The OSC unfortunately makes use of this feature (and
unfortunately can't be fixed because it's a POS), so we're stuck with
this complication.
Implement the worst-case solution and fix this by creating separate ASS
track and renderer objects for each script that wants to set an ASS
overlay.
The z-order is decided by the order the scripts set their text first.
This is essentially random, unless you do it at script init, and you
pass scripts in a specific order. Script initialization is currently
serialized (as a feature), so the first loaded script gets lowest
Z-order.
The Lua script API interestingly remains the same. (And also will remain
undocumented, unsupported, and potentially volatile.)
Instead of passing an explicit cache to the function, the res parameter
is used. Also, instead of replacing its contents, sub bitmaps are now
appended to it (all assuming the format doesn't actually change).
This is preparation for the following commits.
The default of 1.0 was basically making half the algorithm do nothing,
since it turned off all diagonal contributions. The upstream default is
0.6, and this produces a more reasonable image.
The values were changed to reflect an upstream change in the source for
the super-xBR implementation.
The anti-ringing code was basically not working at all, the new
algorithm _significantly_ improves the result (reduces ringing).
This is a fresh implementation from scratch that carries with it
significantly less baggage and verbosity from the previous (ported)
version.
The actual values for the masks and such were copied from the
current code. Behavior and performance should be unaffected.
An important difference between the old code and the new code is that
the new code always explicitly samples from the first component, rather
than being able to process multiple planes at once.
Since prescale-luma only affects luma, I deemed this unnecessary. May
change in the future, if prescale-chroma ever gets implemented. But
prescaling multiple planes would be slow to do this way. (Better would
be to generalize it to differently-sized vectors)
Do not scale OSD mouse input to the ASS OSD script resolution. The
original idea of this mechanism was that the user doesn't have to care
about the actual resolution of anything, and can just use the OSD
resolution consistently. But this made things worse.
Remove the implicit scaling, and always use the screen resolution.
(Except with --vo=xv, where additional scaling is forced upon
everything.)
Drop get_osd_resolution(). There is no replacement. Rename
get_screen_size() and get_screen_margins() to use "osd" instead of
"screen". For anything but --vo=xv these are equivalent, but with
--vo=xv the OSD resolution has additional implicit scaling.
Add code to osc.lua which emulates the old behavior.
Note that none of the changed functions were public API, so implicit
breakage of scripts which used it is just going to happen.
Deselecting cover art and then reselecting it did not work. The second
time the cover art picture is not displayed again. (This seems to break
every other month...)
The reason is commit 6640b22a. It mutates the input packet. And it is
correct that we don't own d_video->header->attached_picture at this
point. Fix it by creating a new packet reference.
Subtitles can be preloaded, which means they're fully read and copied
into ASS_Track. This in turn is mainly for the sake of being able to do
subtitle seeking (when it comes down to it, subtitle seeking is the
cause for most trouble here).
Commit a714f8e92 broke preloaded subtitles which have events with
unknown duration, such as some MicroDVD samples. The event list gets
cleared on every seek, so the property of being preloaded obviously gets
lost.
Fix this by moving most of the preloading logic to dec_sub.c. If the
subtitle list gets cleared, they are not considered preloaded anymore,
and the logic for demuxed subtitles is used.
As another minor thing, preloadeding subtitles did neither disable the
demux stream, nor did it discard packets. Thus you could get queue
overflows in theory (harmless, but annoying). Fix this by explicitly
discarding packets in preloaded mode.
In summary, now the only difference between preloaded and normal
demuxing are:
1. a seek is issued, and all packets are read on start
2. during playback, discard the packets instead of feeding them to the
subtitle decoder
This is still petty annoying. It would be nice if maintaining the
subtitle index (and maybe a subtitle packet cache for instant subtitle
presentation when seeking back) could be maintained in the demuxer
instead. Half of all file formats with interleaved subtitles have
this anyway (mp4, mkv muxed with newer mkvmerge).
Commit 503c6f7f essentially removed timestamps from "laces" (Block sub-
divisions), which means many audio packets will have no timestamp.
There's no reason why bitrate calculation can't just delayed to a point
when the next timestamp is known.
Fixes#2903 (no audio bitrate with mkv files).
Although there is logic to prune subtitles as soon as they get too old
in this mode, this is not done for the _currently_ shown subtitles. Thus
explicitly clearing subtitles on seek is required to avoid duplicate
subtitles in certain cases when seeking.
Instead of hard-coding the logic and planes to skip, factor this out
to a reusible function, and instead add the number of relevant
coordinates to the texture state.
Since prescale now literally only affects the luma plane (and the
filters are all designed for luma-only operation either way), the option
has been renamed and the documentation updated to clarify this.
This is a pretty major rewrite of the internal texture binding
mechanic, which makes it more flexible.
In general, the difference between the old and current approaches is
that now, all texture description is held in a struct img_tex and only
explicitly bound with pass_bind. (Once bound, a texture unit is assumed
to be set in stone and no longer tied to the img_tex)
This approach makes the code inside pass_read_video significantly more
flexible and cuts down on the number of weird special cases and
spaghetti logic.
It also has some improvements, e.g. cutting down greatly on the number
of unnecessary conversion passes inside pass_read_video (which was
previously mostly done to cope with the fact that the alternative would
have resulted in a combinatorial explosion of code complexity).
Some other notable changes (and potential improvements):
- texture expansion is now *always* handled in pass_read_video, and the
colormatrix never does this anymore. (Which means the code could
probably be removed from the colormatrix generation logic, modulo some
other VOs)
- struct fbo_tex now stores both its "physical" and "logical"
(configured) size, which cuts down on the amount of width/height
baggage on some function calls
- vo_opengl can now technically support textures with different bit
depths (e.g. 10 bit luma, 8 bit chroma) - but the APIs it queries
inside img_format.c doesn't export this (nor does ffmpeg support it,
really) so the status quo of using the same tex_mul for all planes is
kept.
- dumb_mode is now only needed because of the indirect_fbo being in the
main rendering pipeline. If we reintroduce p->use_indirect and thread
a transform through the entire program this could be skipped where
unnecessary, allowing for the removal of dumb_mode. But I'm not sure
how to do this in a clean way. (Which is part of why it got introduced
to begin with)
- It would be trivial to resurrect source-shader now (it would just be
one extra 'if' inside pass_read_video).
stream->info can be NULL if it's the cache wrapper. To be fair,
stream->info is considered private API anyway. So don't access it, but
check the URL instead.
Deals with broken mkv subtitle tracks generated by tvheadend. The subs
are srt, but without packet durations.
We need this logic for CCs anyway. CCs in particular will be unaffected
by this change because they are also marked with unknown duration. It
could be that there are actual demuxers outputting CCs - in this case,
we rely on the fact that they don't set a (meaningless) packet duration
(or we'd have to work that around).
Commit 8d4a179c made subtitle decoders pick up fonts strictly from the
same source file (i.e. the same demuxer).
It breaks some fucked up use-case, and 2 people on this earth complained
about the change because of this. Add it back.
This copies all attached fonts on each subtitle init. I considered
converting attachments to use refcounting, but it'd probably be much
more complex.
Since it's slightly harder to get a list of active demuxers with
duplicate removed, the prev_demuxer variable serves as a hack to achieve
almost the same thing, except in weird corner cases. (In which fonts
could be added twice.)
This reverts commit af66fa8fa5.
The reverted commit caused AVCodecContext.channel_layout to be set,
while requesting stereo downmix will make libavcodec output a stupid
message:
ac3: Channel layout '5.1' with 6 channels does not match specified number of channels 2: ignoring specified channel layout
The same happens with --demuxer=lavf (without this change too).
I'm not quite sure what acrobatics are required to shut up libavcodec,
but for now revert the commit. It was a rather minor, almost cosmetic
issue, which I consider less important than clean CLI terminal output.
Completely pointless abominations that FFmpeg refuses to remove. They
are ancient, long deprecated API which we can't use anymore. They
confused users as well.
Pretend that they don't exist. Due to the way --vd works, they can't
even be forced anymore. The older hack which explicitly rejects these
can be dropped as well.
The goal is reducing log messups (which happen surprisingly often) by
buffering partial lines in mp_log. This is still not 100% reliable, but
better.
The extrabuffers for MSGL_STATUS and MSGL_STATS are not needed anymore,
because a separate mp_log instance can be used if problems really occur.
Also, give up, and replace the snprintf acrobatics with bstr.
mp_log.partial has a quite subtle problem wrt. talloc: talloc parents
can not be used, because there's no lock around the internal talloc
structures associated with mp_log. Thus it has to be freed manually,
even if this happens through a talloc destructor.
We want to add a prefix to the ffmpeg log message, so we called mp_msg
multiple times until now. But logging such partial lines is a race
condition, because there's only one internal mp_msg buffer, and no
external mp_msg locks.
Avoid this by building the message on a stack buffer.
I might make a mp_log-local partial line buffer, but even then av_log()
can be called from multiple threads, while targetting the same mp_log.
(Really, ffmpeg's log API needs to be fixed.)
Until now, a rather large stack buffer was used for this, and also a
static buffer in mp_log_root. The latter was added to buffer partial
lines, and the stack buffer was used only for MSGL_STATUS and MSGL_STATS
(I guess because these are the most likely/severe to clash with partial
line buffering).
Make the buffer in mp_log_root dynamically sized, so we don't get cut
off log lines if the text is excessively large. (The OpenGL extension
list dumped by vo_opengl is such an example.)
Since we still have to support partial line buffering (FFmpeg's log
callbacks leave no other choice), keep the stack buffer. But make it
smaller; there's no way all ~6KB are going to be needed in any
situation.
Was only available via --vd=help and --ad=help (i.e. not at all via
client API). Not bothering with separating audio and video codecs, since
this list isn't all that useful anyway in general. If someone complains,
a type field could be added.
Export a number of container fields, which may or may not be useful in
some scenarios. They are explicitly marked as originating from the
demuxer, in order to make it explicit that they might be unreliable.
I'd actually like to remove all other cases where container information
is exported, but those numerous cases are going to be somewhat hard to
deprecate.
Also, not directly related, export the description of the currently
active decoder. (This has been requested before.)
Use the mp_lavc_set_extradata() function instead of setting up the
extradata manually. This takes care of the corner case when
extradata_len is 0.
This apparently fixes#2888.
Hr-seek was often off by one frame due to rounding issues, which have
been traditionally taken care off by adding a "tolerance". Essentially,
frames very close to the seek target PTS are not dropped, even if they
may strictly are before the seek target.
Commit 0af53353 accidentally removed this by always removing frames even
if they're within the "tolerance". Fix this by "unsharing" the logic and
making sure the segment code is inactive for normal seeks.
Ever since a change in mplayer2 or so, relative seeks were translated to
absolute seeks before sending them to the demuxer in most cases. The
only exception in current mpv is DVD seeking.
Remove the SEEK_ABSOLUTE flag; it's not the implied default. SEEK_FACTOR
is kept, because it's sometimes slightly useful for seeking in things
like transport streams. (And maybe mkv files without duration set?)
DVD seeking is terrible because DVD and libdvdnav are terrible, but
mostly because libdvdnav is terrible. libdvdnav does not expose seeking
with seek tables. (Although I know xbmc/kodi use an undocumented API
that is not declared in the headers by dladdr()ing it - I think the
function is dvdnav_jump_to_sector_by_time().) With the current mpv
policy if not giving a shit about DVD, just revert our half-working seek
hacks and always use dvdnav_time_search(). Relative seeking might get
stuck sometimes; in this case --hr-seek=always is recommended.
Adds always-on mode by internally utilizing hidetimeout as negative and
forbidding the user to set negative values.
This removes script-message to enable/disable the osc, and instead introduces a
combined 'visibility' control with the values never/auto/always.
It's available via script_opts and script_message as 'osc-visibility'.
As message, it also supports a 'cycle' value.
The del key is bound to cycling the visibility modes.
There were few issues:
- When it's disabled and then enabled, it was displaying the osc briefly and
then autohide right away. Don't do that.
- When it's enabled and then disabled, it was not removing the osc from screen
if called while the osc is visible (because tick() is responsible for the hide
but it doesn't render() the empty osc when the osc is disabled).
- Due to delayed/async unbinding of mouse events it was possible to show_osc()
after it got disabled e.g. from mouse_move. Prevent this.
This eliminates some intermittent pops heard in a HRT MicroStreamer DAC
uncorrelated with user interaction. As a bonus, this resolves#1773 which I can
o longer reproduce as of this commit. Leave the 50ms buffer for shared mode
since that seems to be working quite well.
This is also the way exclusive mode is done in the MSDN example code:
https://msdn.microsoft.com/en-us/library/windows/desktop/dd370844%28v=vs.85%29.aspx
This was originally increased in c545c40 to mitigate glitches that subsequent
refactorings have eliminated.
A COM message loop is apparently totally inappropriate for a low latency
thread. It leads to audio glitches because the thread doesn't wake up fast
enough when it should. It also causes mysterious correlations between the vo
and ao thread (i.e., toggling fullscreen delays audio feed events). Instead use
an mp_dispatch_queue to set/get volume/mute/session display name from the audio
thread. This has the added benefit of obviating the need to marshal the
associated interfaces from the audio thread.
Don't wait for WASAPI to send another feed event if we detect an underfull
buffer. It seems that WASAPI doesn't always send extra feed events if
something causes rendering to fall behind. This causes every subsequent playback
buffer to under-run until playback is reset. The fix is simply to do a one-shot
double feed when this happens, which allows rendering to catch up with playback.
This was observed to happen when using MsgWaitForMultipleObjects to wait for the
feed event and toggling fullscreen with vo=opengl:backend=win. This commit
improves the behaviour in that specific case and more generally makes exclusive
mode significantly more robust.
This commit also moves the logic to avoid *over*filling the exclusive mode
buffer into thread_feed right next to the above described underfil logic.
OpenSL ES is used on Android. At the moment only stereo output is
supported. Two options are supported: 'frames-per-buffer' and
'sample-rate'. To get better latency the user of libmpv should pass
values obtained from AudioManager.getProperty(PROPERTY_OUTPUT_FRAMES_PER_BUFFER)
and AudioManager.getProperty(PROPERTY_OUTPUT_SAMPLE_RATE).
_Of course_ the previous commit broke --force-window behavior (like it
does every single time I touch it).
vo_has_frame() gets cleared after a seek, so e.g. stopping playback of a
file and going to the next by keeping the seek key down will enter a
short moment without video at the end of the first file, which will set
the stalled_video variable to true. Prevent it by using the indication
whether the window was properly created (which is probably exactly what
we want here).
This function is also responsible for destroying the window when needed,
and obviously we should never do that while video is active. (This is
the actual bug, although the other change in this commit already hides
the common breakage it caused.)
If a stream is marked as EOF (due to no packets found in reach), then we
need to wakeup the decoder. This is important especially if no packets
are found at the start of the file, so the A/V sync logic actually
starts playback, instead of waiting for packets that will never come.
(It would randomly start playback when running the playback loop due to
arbitrary external events like user input.)
Some oddity that is not needed anymore. The only thing which still
referenced them was avoiding loading external files more than once,
which is now prevented by checking the list of tracks instead.
Why was this done so stupidly, with so many complicated special cases,
before? Declare it once so the shader bits don't have to figure out where
and when to do so themselves.
Commit 943f76e6, which already tried this, was very stupid: it didn't
actually override the samplerate for Opus, but overrode it for all
codecs other than Opus. And even then, it failed to use the overridden
samplerate. (Sigh...)
Fixes relative seeks. Without this, a seek back could skip so much data
that the seek would effectively jump forward. (Or insert silence for
files with video.)
There's the question whether the frontend should do this instead (by
using information from the decoders), but for now this seems more
proper.
demux_mkv.c does this already, sort of.
libavformat doesn't for seeks in .ogg (aka .opus), but might be doing it
for mkv. Seems to be a mess as well.
Fixes correctness_trimming_nobeeps.opus. One nasty thing is that this
mechanism interferes with the container-signalled mechanism with
AV_FRAME_DATA_SKIP_SAMPLES. So apply it only if that is apparently not
present. It's a mess, and it's still broken in FFmpeg CLI, so I'm sure
this will get fucked up later again.
I'm not quite sure what the FFmpeg AV_FRAME_DATA_SKIP_SAMPLES API
demands here. The code so far assumed that skipping can be more than a
frame, but not trimming. Extend it to trimming too.
This is actually already done by dec_audio.c. But if
AV_FRAME_DATA_SKIP_SAMPLES is applied, this happens too late here. The
problem is that this will slice off samples, and make it impossible for
later code to reconstruct the timestamp properly.
Missing timestamps can still happen with some demuxers, e.g. demux_mkv.c
with Opus tracks. (Although libavformat interpolates these itself.)
I think the conclusion is that AV_PKT_DATA_SKIP_SAMPLES is misdesigned
(at least for some formats), and an alternative mechanism using
durations would be better. (Combining it with a proper timebase would
keep sample-accuracy.)
This happens only if the new segment wasn't read yet.
This is not quite proper and a problem with dec_sub.c internals.
Ideally, it'd wait with rendering until a new enough segment has been
read. Normally, the new segment is available immediately, so the end
will be automatically clipped by switching to the right segment in the
exact moment it's supposed to become effective.
Usually shouldn't cause any problems, though.
Doing --hwdec=auto ends up picking dxva2, creating a decoder, and then
sending D3D frames down the video chain, which immediately fails and
falls back to software.
Consider dxva2 only if the VO provides a context. If this fails,
autoprobing will proceed to try dxva2-copy as usual.
Fixes#2844.
This is in preparation for a hypothetical API change in libavcodec,
which would allow the decoder to return multiple video frames before
accepting a new input packet.
In theory, the body of the if() added to vd_lavc.c could be replaced
with this code:
packet->buffer += ret;
packet->len -= ret;
but currently this is not needed, as libavformat already outputs one
frame per packet. Also, using libavcodec this way could lead to a
"deadlock" if the decoder refuses to consume e.g. garbage padding, so
enabling this now would introduce bugs.
(Adding this now for easier testing, and for symmetry with the audio
code.)
There is some strange code which sets the DTS of the packet to PTS (but
only if it's not AVI), which apparently helps with timestamp
determination with some broken files. This code is annoying because it
tries to avoid mutating the packet (which it logically doesn't own).
Move it to where it does and get rid of the packet_copy mess.
Needed for the following commit.
This tries to determine whether packet PTS values are accurate and can
be used for frame dropping during seeking. Move both checks (PTS is
missing; PTs is non-monotonic) to the earliest place where they can be
done.
The WGL_NV_DX_interop spec says that a shared IDirect3DSurface9 must not
be lockable, but off-screen plain surfaces are always lockable and using
them causes Nvidia drivers to crash. Use a rendertarget for the shared
surface instead.
This also changes the name of the DX_interop handle for the rendertarget
to match the name of the DirectX object (rather than the GL one) to
match the convention used in context_dxinterop.c.
This file was rewritten from scratch in 0cef033, so it should be okay.
As mentioned in #730, it's a complete rewrite referencing only MSDN and
POSIX, rather than the original code.
Apple crap (namely hardware decoding interop) forces us to use rectangle
textures for input. But after that we continue with normal textures.
This was not considered for debanding, and the sampler type used for it
can be different depending on the exact render chain. Simply use the
target type of the input texture.
* use mp_HRESULT_to_str/mp_LastError_to_str
* make some messages non-identical
* replace "GL" -> "OpenGL"
* change some MP_FATAL to MP_ERR that don't actually kill the vo
This is useful in particular for GetLastError, unfortunately, it's stil pretty
dumb with regards to WASAPI or D3D specific errors, so keep the
hresult_to_string switch.
Apparently, some drivers require you to allocate all of the decoder d3d surfaces
at once. This commit changes the strategy from allocating surfaces as needed via
mp_image_pool_set_allocator, to allocating all the surfaces in one call to
IDirectXVideoDecoderService_CreateSurface and adding them to the pool with
mp_image_pool_add.
fixes#2822
Provide a way for the user to add mp_images to the pool. This is required for
dxva2, for which using set_allocator is extremely awkward since all the d3d9
surfaces must be allocated in advance and all together.
I mistakenly copied the wrong license text into these files when
I created them. Since I'm the only one to have touched these files,
it should be OK to change them.
This is achieved indirectly by deslecting all streams for the non-
current segment (and if the segment doesn't share the demuxer with the
currently active one).
Restores functionality added with commit 46bcdb70.
This uses a different method to piece segments together. The old
approach basically changes to a new file (with a new start offset) any
time a segment ends. This meant waiting for audio/video end on segment
end, and then changing to the new segment all at once. It had a very
weird impact on the playback core, and some things (like truly gapless
segment transitions, or frame backstepping) just didn't work.
The new approach adds the demux_timeline pseudo-demuxer, which presents
an uniform packet stream from the many segments. This is pretty similar
to how ordered chapters are implemented everywhere else. It also reminds
of the FFmpeg concat pseudo-demuxer.
The "pure" version of this approach doesn't work though. Segments can
actually have different codec configurations (different extradata), and
subtitles are most likely broken too. (Subtitles have multiple corner
cases which break the pure stream-concatenation approach completely.)
To counter this, we do two things:
- Reinit the decoder with each segment. We go as far as allowing
concatenating files with completely different codecs for the sake
of EDL (which also uses the timeline infrastructure). A "lighter"
approach would try to make use of decoder mechanism to update e.g.
the extradata, but that seems fragile.
- Clip decoded data to segment boundaries. This is equivalent to
normal playback core mechanisms like hr-seek, but now the playback
core doesn't need to care about these things.
These two mechanisms are equivalent to what happened in the old
implementation, except they don't happen in the playback core anymore.
In other words, the playback core is completely relieved from timeline
implementation details. (Which honestly is exactly what I'm trying to
do here. I don't think ordered chapter behavior deserves improvement,
even if it's bad - but I want to get it out from the playback core.)
There is code duplication between audio and video decoder common code.
This is awful and could be shareable - but this will happen later.
Note that the audio path has some code to clip audio frames for the
purpose of codec preroll/gapless handling, but it's not shared as
sharing it would cause more pain than it would help.
When playback of a video ends, and the next file has no video at all (no
cover art or anything), then the window must be cleared.
This also resizes the window forcibly, which is by design.
Fixes#2825.
A client API user is allowed to call mpv_opengl_cb_uninit_gl() followed
by mpv_opengl_cb_init_gl(). This crashed; fix it by fixing the lifetime
of ctx->gl.
This is required so that the individual surfaces can pass beyond the dxva2
decoder and be passed to the vo.
This also adds additional data to mp_image->planes[0] for IMGFMT_DXVA2, which is
required for maintaining and releasing the surface even if the decoder code is
uninited.
The IDirectXVideoDecoder itself is encapsulated together with its surface pool
and configuration in a dxva2_decoder structure whose creation and destruction is
managed by talloc.
Especially useful to see what video formats are involved on the various
filter links.
I suspect this function is not available on Libav, so add necessary
ifdeffery preemptively.
Don't allow rounding to let it underflow to 0. 0 width or height is
simply not allowed and could cause problems otherwhere.
Indirectly fixes CID 1350057, which complains about not checking the
resulting output size values before using it in divisions.
It thinks that integer_conv_fbo[index] is implied to be accessed with up
to index=5. Although that is theoretical only, it has a point that this
makes no sense. Use the same constant for the array allocation, to make
it more uniform and robust.
Fixes CID 1350060.
The code is shared with the --vd-lavc-threads option, so using 0 for
auto-detection just works.
But no, this is not useful. Just change it for orthogonality.
Until now (and in mplayer traditionally), avi timestamps were handled
with a timestamp FIFO. AVI timestamps are essentially just strictly
increasing frame numbers and are not reordered like normal timestamps.
Limiting the FIFO is required because frames can be dropped. To make
it worse, frame dropping can't be distinguished from the decoder not
returning output due to increasing the buffering required for B-frames.
("Measuring" the buffering at playback start seems like an interesting
idea, but won't work as the buffering could be increased mid-playback.)
Another problem are skipped frames (packets with data, but which do
not contain a video frame).
Besides dropped and skipped frames, there is the problem that we can't
always know the delay. External decoders like MMAL are not going to
tell us. (And later perhaps others, like direct VideoToolbox usage.)
In general, this works not-well enough that I prefer the solution of
passing through AVI timestamps as DTS. This is slightly incorrect,
because most decoders treat DTS as mpeg-style timestamps, which
already include a b-frame delay, and thus will be shifted by a few
frames. This means there will be a problem with A/V sync in some
situations.
Note that the FFmpeg AVI demuxer shifts timestamps by an additional
amount (which increases after the first seek!?!?), which makes the
situation worse. It works well with VfW-muxed Matroska files, though.
On RPI, the first X timestamps are broken until the MMAL decoder "locks
on".
It would make somewhat sense for libcs which don't implement locales at
all, such as Bionic.
Beyond that, setlocale() is specified that it can return NULL, and we
shouldn't crash if that happens.
Unfortunately I see no better solution.
The refresh seek is skipped if the amount of buffered audio is not
overly huge.
Unfortunately softvol af_volume insertion still can cause this issue,
because it's outside of the normal dynamic filter chain changing code.
Move the video refresh call to reinit_video_filters() to make it more
uniform along with the audio code.
The ctx->redrawing field signals whether flip_page() should block. Do
not block if a black frame (i.e. nothing) is to be rendered.
Also, frame==NULL can never happen.
To be more specific, enable vo_opengl and vo_opengl_cb, if libmpv is
compiled, and the GL headers happen to be in the default search paths.
Although platform specific code can be useful for libmpv (for window
embedding, and even with vo_opengl_cb for certain forms of hardware
decoding), it's not a requirement to use the opengl_cb API.
Enabling vo_opengl is not useful here, but the rest of the build system
doesn't distinguish vo_opengl and vo_opengl_cb, and I see no reason to.
We either need to be on Windows, or have posix_spawn available.
If someone can come up with a system that is POSIX, but does not provide
posix_spawn, we could make it optional.
This was dumb. Could make it burn 100% CPU and not exit at the end.
(Because it would retry as instructed, instead of terminating playback.)
It also needs to consider EOF as waiting for input. lavfi_process() will
decide if it's really EOF, or if further input might come in the future.
Without this, it'd would think that it does not need to wait for input,
i.e. that new input will be available immediately.
(Not so fond of the duplication of subtle logic.)
fd339e3f53 introduced a regression that caused
segfault while uniniting dxva2 decoder (and possibly vdpau too). The problem was
that it freed the avctx earlier, before calling the backend-specific uninit
which referenced it.
Revert some of the changes of that commit, and avoid calling flush by
checking whether the codec is open instead.
(Based on a PR by Kevin Mitchell.)
Signed-off-by: wm4 <wm4@nowhere>
It doesn't provide this function. The code is not really designed to
work without it, so it will probably mess up big time, but at least
make it compile again.
See --lavfi-complex option.
This is still quite rough. There's no support for dynamic configuration
of any kind. There are probably corner cases where playback might freeze
or burn 100% CPU (due to dataflow problems when interaction with
libavfilter).
Future possible plans might include:
- freely switch tracks by providing some sort of default track graph
label
- automatically enabling audio visualization
- automatically mix audio or stack video when multiple tracks are
selected at once (similar to how multiple sub tracks can be selected)
The complex filter support that will be added makes much more complex
use of libavfilter, and I'm not going to bother with adding hacks to
keep libavfilter optional.
track can't be NLUL at this point, so the if is redundant. Remove it and
unindent the block. Also, make the function check whether the track is
selected at all, which makes it safer and idempotent.
This makes it possible to set video size and position using the
--geometry and/or --autofit options. It's also possible to switch
between fullscreen/non-fullscreen playback during runtime.
It can be "dangerous". In particular, the decoder might have failed to
initialize, and is now in a broken state. avcodec_flush_buffers() is not
expected to be called in this state, and could trigger undefined
behavior.
I can't explain this, but it seems to be a similar case to the ALSA HDMI
one. I find it hard to tell because of the slightly different names and
conventions in use in libavcodec, WAVEEXT channel masks, decoders, codec
specifications, HDMI, and platform audio APIs.
The fix is the same as the one for ao_alsa (see commit be49da72). This
should fix at least playing 7.1 sources on OSX with 7.1(rear) selected
in Audio MIDI Setup. The ao_alsa commit mentions XBMC, but I couldn't
find out where it does that or if it also does that for CoreAudio. It's
woth noting that PHT (essentially an old XBMC fork) also exhibited the
incorrect behavior (i.e. side and back speakers were swapped).
Remove flc-frc <-> sl<->sr. This was just plain wrong, and a mistaken
change to make 7.1 work properly on CoreAudio with 7.1(rear) layout.
Also see the following commit.
Add br-br <-> sl<->sr, because we decided that it makes sense.
Note that this "fudging" is applied only if the channel pairs are
replaced, i.e. they would get dropped and be replaced with silence. This
is done to compensate for libswresample's default rematrixing (which
takes care of some more common cases).
Avoids "problems". In particular, it makes MMAL output a NOPTS timestamp
if the input timestamp was NOPTS.
Don't do it for other decoders. Ideally, we will at some point in the
future switch to integer fractions for timestamps at least up until the
filter layer. But this would be a larger change, and for now I'd prefer
keeping the not-rounded demuxer timestamps (if we have them).
This is the "unicode" version of it. It appears Firefox uses it now?
I'm not sure if we still need to support the old variant, but hopefully
not.
Fixes#2782.
Will be helpful for the coming filter support. I planned on merging
audio/video decoding, but this will have to wait a bit longer, so only
remove the duplicate status codes.
Now I'm not sure if this is really necessary to explicitly mention. But
then there have been issues and pull requests without being properly
looked at for a week or longer.
Since there can be multiple backends for a single API (vaapi can use GLX
or EGL), not logging the exact backend name is annoying. So add it. At
the same time, there is no need to duplicate the name as used by the
--hwdec options, so replace it with using the numeric hwdec API ID.
Let's fix broken samples with questionable heuristic without real
reasoning. Until this gets fixed properly, this is a good compromise,
though. A proper fix would properly resync audio and video without
brutally resetting the decoders, but on the other hand not doing the
brutal reset would cause issues in other obscure corner cases such
resyncing might cause.
This code is tricky because it has to wakeup the mainloop to make
progressing during syncing audio, but also has to avoid waking it up
when it's not needed. Failure to do so either burns CPU by not ever
going to sleep, or causes apparent "freezes" by going to sleep (and it
will continue if the mainloop is woken up e.g. due to user input).
In this case, simply starting A/V playback with --start=5 and removing
an unrelated wakeup in osd.c can trigger such a "freeze". The unrelated
wakeup did hide this bug, nonetheless it's a bug.
(Can't wait to rewrite this shitty audio resync code. And it's all my
fault.)
Commit b53cb8de added a delay queue for decoded frames. This is supposed
to be used with copy-back decoders like dxva2-copy and vaapi-copy.
Surfaces returned by them can't be referenced after uninitializing the
decoders, so they have to be released before destroying the decoder.
Move the flush_all() call above decoder uninit accordingly. Also move
the destruction of the AVFrame used for decoding (just for being
defensive - normally it doesn't hold any reference).
We just need to provide an entrypoint for it, and move the main init
code to a separate function. This gets rid of the messy video chain full
reinit in command.c, which completely destroyed and recreated the video
state for the purpose of mid-stream hw/sw switching.
Don't give the "software_fallback_decoder" field special meaning. Alwass
set it, and rename it to "decoder". Whether hw decoding is used is
determined by the "hwdec" field already.
These changes don't make too much sense without context, but are
preparation for later. Then the audio_src/video_src fields will be
actually be NULL under circumstances.
Before this commit, reinit_audio_chain() did 2 things: create all the
management data structures and initialize the decoder, and handling lazy
filter/output init (as well as dealing with format changes). For the
second purpose, it could be called multiple times (even though it wasn't
really idempotent). This was pretty weird, so make them separate
functions. The new function is actually idempotent too.
It also turns out the reinit functions don't have to call themselves
recursively for the spdif PCM fallback.
Regression caused by commit 3b95dd47. Also see commit 4c25b000. We can
either use video_next_pts and add "delay", or we just use video_pts. Any
other combination breaks. The reason why the assumption that delay==0 at
this point was wrong exactly because after displaying the first video
frame (usually done before audio resync) a new frame might be "added"
immediately, resulting in a new video_next_pts and "delay", which will
still amount to video_pts.
Fixes#2770. (The reason why display-sync was blamed in this issue is
because enabling display-sync in the options forces a prefetch by 2
instead of 1 frames for seeks/playback restart, which triggers the
issue, even if display-sync is not actually enabled. In this case,
display-sync is never enabled because the frames have a unusually high
frame duration. This is also what exposed the initial desync issue.)
This codes tries to deal with broken PTS timestamps, but since commit
271cabe6 it didn't always overwrite the previous timestamp as it should
have. This mattered only if there were broken timestamps in the video
stream.
Also remove the pointless prev_codec_pts variables, since the decoder
doesn't overwrite these fields anymore.
This also makes it refcounted, i.e. the new AVFrame will reference the
mp_audio buffers, instead of potentially forcing the consumer of the
AVFrame to copy the data.
All the extra code is for handling the >8 channels case, which requires
very messy dealing with the extended_ fields (not our fault).
Correctly avoid a reload if the current device was specified by the user through
--audio-device. Previously, we only recognized if the user had specified
--ao=wasapi:device=.
FFmpeg can generate such files. It's unclear whether they're allowed by
Matroska. mkvinfo shows packet timestamps in both forms (one of them
must be a bug), and at last libavformat's demuxer treats timestamps
as signed.
GLES requires this. Some more common sampler types have default
precisions, but not usampler2D. Newer ANGLE builds verify this more
strictly than older builds, so this wasn't caught before.
Fixes#2761.
GLES does not support high bit depth fixed point textures for unknown
reasons, so direct 10 bit input is not possible. But we can still use
integer textures, which are supported by GLES 3.0. These store integer
data just like the standard fixed point textures, except they are not
normalized on sampling. They also don't support bilinear filtering, and
require a special sampler ("usampler2D").
While these texture formats enable us to shuffle the data to the GPU,
they're rather impractical with the requirements mentioned above and our
current architecture. One problem is that most code assumes it can
always use bilinear scaling (even if bilinear is never used when using
appropriate scale/cscale options). Another is that we don't have any
concept of running a function on a texture in an uniform way.
So for now, run a simple conversion step through a FBO. The FBO will use
the rgba16f format normally, which gives enough bits for 10 bit, and
will at least gracefully degrade with higher depth input.
This is bound to be much slower than a more "direct" method, but at
least it works and is simple to implement.
The odd change of function call order in init_video() is to properly
disable "dumb mode" (no FBO use) if these texture formats are in use.
This was never reset - absolutely can't be right. If the renderer
somehow switches back to another codepath, it certainly has to be reset.
Maybe this was hard to hit, as the normalization is going to be
idempotent in simpler cases (like rendering RGBA input).
Also get rid of the "merged" variable.
This seems generally easier when using libmpv (and was already requested
and implemented before: see commit 327a779a; it was reverted some time
later).
With the weird internal logic we have to deal with, in particular the
--softvol=no case (using system volume), and using the audio API's mixer
(--softvol=auto on some systems), we still can't avoid all glitches and
corner cases that complicate this issue so much. The API user is either
recommended to use --softvol=yes or auto, or to watch the new
mixer-active property, and assume the volume/mute properties have
significant values if the mixer is active.
Remaining glitches:
- changing the volume/mute properties has no effect if no internal mixer
is used (--softvol=no) and the mixer is not active; the actual mixer
controls do not change, only the property values
- --volume/--mute do not have an effect on the volume/mute properties
before mixer initialization (the options strictly are only applied
during mixer init)
- volume-max is 100 while the mixer is not active
Commit b53cb8de increased this by the number of additionally delayed
surfaces. But since this is only enabled in copy-back mode (which is
what process_image is about), the other additional surfaces accounted
for the direct rendering case can be ignored.
Often requested. The main argument, that prominent scalers like sharpen
change the image even if no scaling happens, disappeared anyway.
("sharpen", unsharp masking, is neither prominent nor a scaler anymore.
This is an artifact from MPlayer, which fuses unsharp masking with
bilinear scaling in order to make it single-pass, or so.)
Another fix for the crazy and insane nvidia preemption behavior.
This time, the situation is that we are using vo_opengl with vdpau
interop, and that vdpau got preempted in the background while mpv was
sitting idly. This can be e.g. reproduced by using:
--force-window=immediate --idle --hwdec=vdpau
and switching VTs. Then after switching back, load a video file.
This will not let mp_vdpau_handle_preemption() perform preemption
recovery, simply because it will do so only once vdp_decoder_create()
has been called. There are some other API calls which trigger
preemption, but many don't.
Due to the way the libavcodec API works, vdp_decoder_create() is way too
late. It does so when get_format returns. It notices creating the
decoder fails, and continues calling get_format without the vdpau
format. We could perhaps force it to reinit again (by adding a call to
vdpau.c, that checks for preemption, and sets hwdec_request_reinit), but
this seems too much of a mess.
Solve it by calling API in mp_vdpau_handle_preemption() that empirically
does trigger preemption: output_surface_put_bits_native(). This call is
useless, and in fact should be doing nothing (empty update VdpRect).
There's the slight chance that in theory it will slow down operation,
but in practice it's bound to be harmless. It's the likely cheapest and
simplest API call I've found that can trigger the fallback this way.
(The driver is closed source, so it was up to trial & error.)
Also, when initializing decoding, allow initial preemption recovery,
which is needed to pass the test mention above.
With the format left untouched, this would just try to reinit with a
spdif format again.
We're not clearing the format in reset_audio_state() so the audio chain
can be recreated any time without having to wait for a frame to be
decoded.
Some VOs had support for these - remove them.
Typically, these formats will have only some use in cases where using
RGB software conversion with libswscale is faster than letting the
VO/GPU do it (i.e. almost never). For the sake of testing this case,
keep IMGFMT_RGB565. This is the least messy format, because it has no
padding/alpha bits with unknown semantics.
Note that decoding to these formats still works. We'll let libswscale
repack the data to whatever the VO in use can take.
Notably, the address of the enumerator->count member is passed to
IMMDeviceCollection::GetCount(), which expects a UINT variable, not an int. How
did this ever work?
Previously, if the enumerator found no devices, attempting to get the default
device with IMMDeviceEnumerator::GetDefaultAudioEndpoint would result in the
cryptic (and undocumented) E_PROP_ID_UNSUPPORTED. This way, the user is given a
better indication of what exactly is wrong and isolates any other possible
triggers for this error.
At least DTV_ENUM_DELSYS is not available in older versions.
It's hard to tell when this identifier was introduced, but it appears it
was probably API version 5.5.
Even though the timing logic is correct, it tends to mess with looping
videos and such in unappreciated ways.
It also has to be admitted that most file formats seem not to properly
define the duration of the last video frame (or libavformat does not
export it in a useful way), so whether or not we should use the demuxer
reported framerate for the last frame is questionable. (Still, why would
you essentially just discard the last frame?)
The timing logic is kept, but disabled for video with "normal" FPS
values. In particular, we want to keep it for displaying images, which
implicitly set the frame duration to 1 second by reporting 1 FPS. It's
also good for slide shows with mf://.
Fixes#2745.
Most text subtitles are read completely on loading (libavformat works
this way, and there are good reasons to do it on the higher levels too).
This leads to some messy problems. For example, the subtitle path is the
only one which might read packets during decoder initialization. This is
not neccessary; get rid of it.
This fixes a potential problem of seeking to position 0 for image
subtitles on init, and if the start position is not at the beginning of
the timeline.
It doesn't need to be part of the big context, but is strictly part of
shuffling data from the audio filters to audio output, and thus belongs
into ao_chain.
It also turns out that clearing it in clear_audio_output_buffers() is
completely redundant.
(Of course ao_buffer is an abomination in the first place and shouldn't
exist at all.)
vo_chain_uninit() isn't supposed to care much about the decoder
(although decoders and outputs still go strictly together, so there is
not much of an actual difference now).
Also unset track.d_video correctly.
Remove a stale declaration from dec_video.h as well.
Similar to the video path. dec_audio.c now handles decoding only. It
also looks very similar to dec_video.c, and actually contains some of
the rewritten code from it. (A further goal might be unifying the
decoders, I guess.)
High potential for regressions.
Using the new API is a necessity for multiple-delivery-system
devices, since the old API does not offer a way to switch
the delivery system of the card.
This should in principle also be done for DVB-T / ATSC,
especially since most DVB-T devices also support DVB-C,
but I can not test such an implementation due to lack of hardware
(currently) so it seems better to leave the existing, tested code-path
in place for now.
No need use use all capital letters, and don't warn
if DVB-S2 is supported in addition since we handle that
in DVB-S case already.
Also, print the delivery system number for still unhandled
delivery systems to simplify debugging.
Saves one unnecessary additional ioctl per tuning
by just reusing existing information.
Should also fix the case of multiple supported delivery types
since we now rely on the initial query from the chosen
configuration after channel list parsing
instead of requerying the device.
Most common case would be DVB-C / DVB-T combination cards.
Cards with multiple delivery systems are only supported
starting from DVBv5 API (Kernel 2.6.38).
In this case, we loop over all delivery systems and
just treat them as different cards would be treated:
They all get their own TUNER-type, channel-list parsing etc.
Only request the current screen configuration instead of polling for new
screens, too. We're not interested in detecting any new screens as we're
merely enumerating what is currently connected and configured.
On some hardware (like mine) calling XRRGetScreenResources will stall
X11 for about 10 to 20 seconds. This has annoyed me for a few months
now and almost made me switch to VLC ;)
Signed-off-by: wm4 <wm4@nowhere>
This is probably the 3rd time the user-visible behavior changes. This
time, switch back because not normalizing seems to be the more expected
behavior from users.
Too many problems. Well, actually it's just Linux audio systems which
cause problems, and exclusive audio access on other platforms.
In any case, it seems you have to do some manual configuration if you
want multichannel audio output.
Seems useless.
This only helped in one case: one audio stream in the sample
av_find_best_stream_fails.ts had a AC3 packets which couldn't be
decoded, and for which avcodec_decode_audio4() returned 0 forever. In
this specific case, playback will now not start, and you have to
deselect audio manually.
(If someone complains, the old behavior might be restored, but
differently.)
Also remove the stale "bitrate" field.
While the situation is not really clear for the other rewritten
coreaudio code, it's very clear for the channel mapping code. It was all
written by us. (MPlayer doesn't even have any channel map handling.)
Do this to make the license situation less confusing.
This change should be of no consequence, since LGPL is compatible with
GPL anyway, and making it LGPL-only does not restrict the use with GPL
code.
Additionally, the wording implies that this is allowed, and that we can
just remove the GPL part.
This covers source files which were added in mplayer2 and mpv times
only, and where all code is covered by LGPL relicensing agreements.
There are probably more files to which this applies, but I'm being
conservative here.
A file named ao_sdl.c exists in MPlayer too, but the mpv one is a
complete rewrite, and was added some time after the original ao_sdl.c
was removed. The same applies to vo_sdl.c, for which the SDL2 API is
radically different in addition (MPlayer supports SDL 1.2 only).
common.c contains only code written by me. But common.h is a strange
case: although it originally was named mp_common.h and exists in MPlayer
too, by now it contains only definitions written by uau and me. The
exceptions are the CONTROL_ defines - thus not changing the license of
common.h yet.
codec_tags.c contained once large tables generated from MPlayer's
codecs.conf, but all of these tables were removed.
From demux_playlist.c I'm removing a code fragment from someone who was
not asked; this probably could be done later (see commit 15dccc37).
misc.c is a bit complicated to reason about (it was split off mplayer.c
and thus contains random functions out of this file), but actually all
functions have been added post-MPlayer. Except get_relative_time(),
which was written by uau, but looks similar to 3 different versions of
something similar in each of the Unix/win32/OSX timer source files. I'm
not sure what that means in regards to copyright, so I've just moved it
into another still-GPL source file for now.
screenshot.c once had some minor parts of MPlayer's vf_screenshot.c, but
they're all gone.
Previously used opt_exclusive option to decide which volume control code to run.
The might not always reflect the actual state, for example if passthrough
is used. Admittedly, none of the volume controls will work anyway with
passthrough, but this is the right thing to do.
Slightly helps with timeline stuff, like EDL. There is no need to keep
network (or even just disk I/O) busy for all segments at the same time,
because 1. the data won't be needed any time soon, and 2. will probably
be discarded anyway if the stream is seeked when segment is resumed.
Partially fixes#2692.
Prevents channels from being dropped, e.g. when going 7.1 -> 7.1(wide)
and similar cases. The reasoning here is that channel layouts over HDMI
don't work anyway, and not dropping a channel and playing it on a
slightly "wrong" (but expected) speaker is preferable to playing silence
on these speakers.
Do this to remove issues with ao_coreaudio. Frankly I'm not sure whether
our mapping (between CA and mpv/FFmpeg speakers) is correct, but on the
other hand due to the reasons stated above it's not all that meaningful.
demux_lavf.c leaked the complete subtitle data if it was put through
iconv.
lavc_conv.c leaked AVCodecContext.subtitle_header (set by libavcodec),
which is fixed by using avcodec_free_context(). It also leaked the
subtitle that was decoded last.
Eventually we want the VO be driven by a A->V filter, so a decoder
doesn't even have to exist. Some features definitely require a decoder
though (like reporting the decoder in use, hardware decoding, etc.), so
for each thing which accessed d_video, it has to be redecided if and how
it can access decoder state.
At least the "framedrop" property slightly changes semantics: you can
now always set this property, even if no video is active.
Some untested changes in this commit, but our bio-based distributed
test suite has to take care of this.
This moves some code related to decoding from video.c to dec_video.c,
and also removes some accesses to dec_video.c from the filtering code.
dec_video.ch is starting to make sense, and simply returns video frames
from a demuxer stream. The API exposed is also somewhat intended to be
easily changeable to move decoding to a separate thread, if we ever want
this (due to libavcodec already being threaded, I don't see much of a
reason, but it might still be helpful).
Makes the next commit simpler. It's probably a bad idea to add more
fields to the global state, but on the other hand the client API state
is pretty much per-instance anyway. It also will help with things like
the proposed libmpv custom stream API.
The aspect ratio calculations are cached (mainly so that aspect ratio
related messages are not logged on every frame). The cache is not clared
anymore when video filters are reconfigured, but changing the
video-aspect-ratio property relied on it. Make it explicit.
Fixes#2714.
Since the streams are chosen from the full TS by the player frontend,
one should not expect that the program which is shown matches the chosen
channel which was used for tuning to the frequency.
Also, reformulate slightly to simplify reading.
This is labeled as "name of the service in broadcasting (channel name)"
and exported as a generic tag by avformat-demuxers, notably lavf
when handling mpegts-streams from DVB.
The binding is similar to the tv-binding, just with capital letters.
Switching the dvb-channel-name property compared to dvb-channel
means the channel-name is shown on-screen when switching instead of
"dvb-channel (error)" otherwise,
and switching anyways happens without changing the card.
Channel switching is treated inside the global DVB state
by now. Anyways the last switching direction is not really useful
and of no interest inside the player.
On read, it returns the name of the current DVB program,
on write, it triggers a channel-switch to the program
if it is found in the channel list of the currently active card.
Compared to the dvb-channel property which already exists
and is a pair of integers (card + channel number) this has the limitation
of not switching the card, but is probably of much more common use.
The mutex is used in dvbin_open and dvbin_close only since these are
the only entry / exit points to the stream.
When opening, it is first checked (mutexed) whether the state already exists
and is in use, then a STREAM_ERROR is returned,
since there may be only one stream_dvb active at a time.
State-creation itself is also protected by mutex.
In dvbin_close, the usage-bit is set to false (mutexed) in case
of channel switch.
In case of stream-teardown, the state is destructed
(also protected by mutex).
The state-structure is kept in a static pointer and reused on
recreation of the stream.
To not leak the state and the FDs within upon mpv shutdown,
the state-structure is still destructed gracefully in dvbin_close(),
unless a channel switch has been initiated directly before.
This fixes channel-switching for DVB which was broken since a609877.
The state-struct now contains everything which can be kept after initial initialization.
This includes the channel-lists, configuration, device-fds and also information like
current channel and current card.
The dvb_priv_t is kept containing the mp-options, a pointer to the state and to the logger.
After this restructuring, the state-struct contains all information which can be persisted
across channel switching.
Lots of noise to remove the vfilter/vo fields from dec_video.
From now on, video filtering and output will still be done together,
summarized under struct vo_chain.
There is the question where exactly the vf_chain should go in such a
decoupled architecture. The end goal is being able to place a "complex"
filter between video decoders and output (which will culminate in
natural integration of A->V filters for natural integration of
libavfilter audio visualizations). The vf_chain is still useful for
"final" processing, such as format conversions and deinterlacing. Also,
there's only 1 VO and 1 --vf option. So having 1 vf_chain for a VO seems
ideal, since otherwise there would be no natural way to handle all these
existing options and mechanisms.
There is still some work required to truly decouple decoding.
Instead of handling this on filter chain reinit, do it directly after
the decoder. This makes the code less entangled. In particular, this
gets rid of the really weird "override params" concept in the video
filter code.
The last_format/fixed_formats have some redundance with decoder_output,
but unfortunately the latter has a slightly different use.
UTF-16 subtitles are special in that they are usually read by
libavformat directly, even though they are not in UTF-8. This is
explicitly handled convert_charset() and skips conversion to UTF-8.
There was a bug due to not resetting the file position: if conversion
happens, the actual stream is replaced with a memory stream containing
the converted data, but if conversion is skipped, the original stream
with the wrong file position is kept.
Fix by always opening a memory stream. (We _could_ seek back, but there
is a slight possibility of additional failure due to unseekable
streams.)
Also, don't enter conversion if the subtitle is detected as UTF-8
either.
Fixes#2700.
Basically reimplement it. The old implementation was quite stupid, and
was probably done this way because video filtering and output used to be
way less decoupled. Now we can reimplement it in a very simple way: when
backstepping, seek to current time, but keep the last frame that was
supposed to be discarded when reaching the target time. When the seek
finishes, prepend the saved frame to the video frame queue.
A disadvantage is that the new implementation fails to skip over
timeline boundaries (ordered chapters etc.), but this never worked
properly anyway. It's possible that this will be fixed some time in the
future.
This is mainly a refactor. I'm hoping it will make some things easier
in the future due to cleanly separating codec metadata and stream
metadata.
Also, declare that the "codec" field can not be NULL anymore. demux.c
will set it to "" if it's NULL when added. This gets rid of a corner
case everything had to handle, but which rarely happened.
pyqtgraph's intColor() is less than ideal, especially on white
background. Can't see anything.
Unfortunately the rendering of the legend can't be fixed, because
pyqtgraph is terrible and hardcodes its rendering, including colors.
This slightly changes behavior when seeking with external audio/subtitle
tracks if transport streams and mpeg files are played, as well as
behavior when seeking with such external tracks.
get_main_demux_pts() is evil because it always blocks on the demuxer (if
there isn't already a packet queued). Thus it could lock up the player,
which is a shame because all other possible causes have been removed.
The reduced "precision" when seeking in the ts/mpeg cases (where
SEEK_FACTOR is used, resulting in byte seeks instead of timestamp seeks)
might lead to issues. We should probably drop this heuristic. (It was
introduced because there is no other way to seek in files with PTS
resets with libavformat, but its value is still questionable.)
Note that hresult_to_str() (coming from wasapi_explain_err()) is mostly
wasapi-specific, but since HRESULT error codes are unique, it can be
extended for any other use.
This is another attempt at making files with sparse video frames work
better.
The problem is that you generally can't know whether a jump in video
timestamps is just a (very) long video frame, or a timestamp reset. Due
to the existence of files with sparse video frames (new frame only every
few seconds or longer), every heuristic will be arbitrary (in general,
at least).
But we can use the fact that if video is continuous, audio should also
be continuous. Audio discontinuities can be easily detected, and if that
happens, reset some of the playback state.
The way the playback state is reset is rather radical (resets decoders
as well), but it's just better not to cause too much obscure stuff to
happen here. If the A/V sync code were to be rewritten, it should
probably strictly use PTS values (not this strange time_frame/delay
stuff), which would make it much easier to detect such situations and
to react to them.
mp_parse_escape() is used by the JSON parser in json.c, and JSON allows
escaping "/" (solidus).
Although it makes no sense, apparently Javascript traditionally allowed
that as escape sequence for working around issues with embedding
Javascript in HTML. (Or something like this must have been the history
of this issue.) Since it's valid in Javascript, it had to be valid in
JSON as well, and JSON explicitly specifies it as valid escape.
Fixes#2694.
AVComponentDescriptor.offset was introduced relatively recently. On
older releases, you have to use AVComponentDescriptor.offset_plus1,
which is now deprecated.
Instead of adding ifdeffery, assume AV_PIX_FMT_NV21 is the only format
for which this applies (and will remain the only case), which is
probably true enough.
Should take care of the planned FFmpeg AV_PIX_FMT_P010 addition. (This
will eventually be needed when doing HEVC Main 10 decoding with DXVA2
copyback.)
A format could declare that some or all LSBs in a component are padding
bits by setting a non-0 AVComponentDescriptor.shift value. This means we
would interpret it incorrectly, because until now we always assumed all
regular formats have the padding in the MSBs.
Not a single format that does this actually exists, though. But a NV12
variant will be added later in FFmpeg.
Windows definitely supports Unix-style fd inheritance. This mostly
worked when launched from mpv.exe, though mpv should change the file
mode to O_BINARY. When launched from mpv.com, the wrapper must pass the
list of handles (stored in the undocumented lpReserved2 and cbReserved2
fields) to the mpv process.
Commit 3909e4cd ended up losing the ability to tune the gaussian window,
which this commit trivially reintroduces.
The constant scaling factor (present in the code copied from glumpy)
also goes against filter_kernels.c conventions, which is that f(0.0) = 1
(and the invoking code takes care of normalization), and has been
removed.
The values of this new gaussian function corresponds to different
functions when compared against the old version. To translate the old
values p1 to the new values p2 requires solving 2^(e/p1) = e^(2/p2) or
p2 = p1 * 2/(e * ln(2)) ≈ p1 * 1.0615
In other words, to get the old default in the new function requires
setting scale-param1 to 1.0615. (The new function is *slightly* sharper
by default)
(Though most users should probably not notice the change)
To get a uniform license for this file, relicense the mpv parts to BSD
as well.
But leave the door open for a later change to LGPL. (All non-Glumpy code
was written within mpv, and all mpv authors have agreed to LGPL
relicensing.)
Closes#2688.
This file claims to be based on the "MPlayer VA-API patch", but this is
untrue. Only some glue code was copied from hwdec_vaglx.c, and this glue
code was never in MPlayer or the MPlayer VA-API patch in any form, and
instead part of the mpv-original way we do hardware decoding OpenGL
interop. The EGL interop method didn't exist at the time the MPlayer
VA-API patch was created either.
PT_RELOAD_FILE is a somewhat obscure case when using DVB or when
switching Matroska editions. Both cases were broken, because the
asynchronous playback abort mechanism was still triggered. This
mechanism is used to force the demuxer and stream layers to exit
immediately (instead of blocking on I/O possibly forever), and
is normally disabled on playback start. The reopen path is a bit
strange, and needs to reset it manually.
Pointed out in #2568.
If you do "mpv /bla/", and then branch out into sub-directories using
playlist navigation, and then used quit and watch later, then playing
the same directory did not resume from the previous point. This was
because resuming is based on the path hash, so a path prefix can't be
detected when resuming the parent directory.
Solve this by writing each path prefix when playing directories is
involved. (This includes all parent paths, so interestingly, "mpv /"
would also resume in the above example.)
Something like this was requested multiple times, and I want it too.
This commit replaces code based on AGG, taken from this source file:
http://vector-agg.cvs.sourceforge.net/viewvc/vector-agg/agg-2.5/include/agg_image_filters.h
The intention is that filter_kernels.c can be relicensed to LGPL or BSD.
Because the AGG author died, full replacement is the only way to achieve
it.
This affects only some filter functions. These are exclusively
mathematical functions for computing filter coefficients. (Other parts
in filter_kernel.c were originally written by me, with heavy additions
and refactoring done by other mpv contributors.) While the code is
mostly just well-known mathematical formulas written down in C form,
AGG copyright could perhaps be claimed anyway.
To remove the AGG code, I replaced it with the filter functions from:
https://github.com/glumpy/glumpy/blob/master/glumpy/library/build-spatial-filters.py
These functions conveniently compute exactly the same thing in mpv,
Glumpy, AGG (and about anything that will filter images using the same
mathematical principles).
First I ported the Python code in the file to C. Then I replaced all
functions in filter_kernels.c with this code that could be replaced.
Then I investigated whether the remaining functions were based on AGG
code and took appropriate action:
hanning(), hamming(), quadric(), bicubic(), kaiser(), blackman(),
spline16(), spline36(), gaussian(), sinc() were taken straight from
Glumpy.
For sinc(), re-add the "fabs(x) < 1e-8" check, which was added in commit
586dc557 for unknown reasons.
gaussian() loses its filter parameter for some reason. (Well, who cares,
not my problem.)
The really awkward thing is that the text for hanning() and hamming()
does not change. In theory these functions are now based on Glumpy code,
but it seems like this can be neither proven nor denied. (The same
happened in some other cases with at least a few lines of code.)
sphinx() was added in commit 586dc557, and looks suspiciously like
sinc() as well. Replace the first 3 lines of the body with the ported
function (of which 2 lines do not change; the first uses code only in
mpv, and the second is just "return 1.0;"). The 4th line is only similar
on an abstract level (and that because of the mathematical relation
between these functions). Although the original sinc() was probably used
as template for it, with the other lines replaced, I don't think you
could make the claim that it falls under AGG copyright.
jinc() was added in commit 26baf5b9, but the code for it might be based
on sinc(). Rewrite it based on the "new" sinc(). Some of the same
remarks as with sphinx() apply.
cubic_bc() was ported from Glumpy's Mitchell(). (As far as I'm aware,
with the default parameters it's called "the" Mitchell-Netravali filter,
but in mpv this function is used to generate a whole group of filters.)
spline64() was added in commit a8b67c66, and was probably derived from
spline36(). Re-derive it from the "new" spline36().
triangle() could be considered derived from the original bilinear().
This is this in the original commit:
static double bilinear(kernel *k, double x)
{
return 1.0 - x;
}
This _might_ be based on AGG's image_filter_bilinear:
struct image_filter_bilinear
{
static double radius() { return 1.0; }
static double calc_weight(double x)
{
return 1.0 - x;
}
};
Considering that the "framework" was written by me, and the only part
from AGG taken is "return 1.0 - x;", and this part is trivial and was
later thoroughly replaced, this is probably not under the AGG copyright.
I'm hoping this doesn't introduce regressions. But the main focus is not
being productive anyway, and I didn't rigorously check unintended
changes in functionality.
It existed for XP-compatibility only. There was also a time where
ao_wasapi caused issues, but we're relatively confident that ao_wasapi
works better or at least as good as ao_dsound on Windows Vista and
later.
Normally, PulseAudio accepts any combination of sample format, sample
rate, channel count/map. Sometimes it does not. For example, the channel
rate or channel count have fixed maximum values. We should not fail
fatally in such cases, but attempt to fall back to a working format.
We could just send pass an "unset" format to Pulse, but this is not too
attractive. Pulse could use a format which we do not support, and also
doing so much for an obscure corner case is not reasonable. So just pick
a format that is very likely supported.
This still could fail at runtime (the stream could fail instead of going
to the ready state), but this sounds also too complicated. In
particular, it doesn't look like pulse will tell us the cause of the
stream failure. (Or maybe it does - but I didn't find anything.)
Last but not least, our fallback could be less dumb, and e.g. try to fix
only one of samplerate or channel count first to reduce the loss, but
this is also not particularly worthy the effort.
Fixes#2654.
Apparently, the firmware will ignore pixel_x/pixel_y if the numeric
value of them gets too high (even if they indicate square pixel aspect
ratio). Even worse, the destination rectangle is ignored completely,
and the video frame is simply stretched to the screen. I suspect this
is an overflow or weird sanity check within the firmware.
Work it around by limiting the fields to 16000, which is an arbitrary
but apparently working limit.
There are a lot of incorrectly encoded subtitles with .ass extension
and non-ass subtitles (srt, ssa) with such extension, so we need to
try codepage detection even for .ass.
Signed-off-by: wm4 <wm4@nowhere>
Given 5.1(side), this lets it pick 5.1 from [5.1, 7.1]. Which was
probably the original intention of this replacement stuff. Until now,
the opposite was done in some cases.
Keep the old heuristic if the replacement is not perfect. This would
mean that a subset of the channel layout is an inexact equivalent, but
not all of it.
(My conclusion is that audio output APIs should be designed to simply
take any channel layout, like the PulseAudio API does.)
Unify and clean up listing and selection. Use common enumerator code for both
operations to avoid duplication or inconsistencies.
Maintain, but significatnly simplify manual device selection by id, name or
number. This actually fixes loading by name which didn't really work before
since the "name" displayed by --audio-device=help differed from that used to
match the selection, which used the device "description" instead.
Save the selected deviceID in the private structure for later loading. This will
permit moving the device selection into the main thread in a future commit.
GLSL in GLES 2.0 did not have line continuation in its preprocessor.
This broke shader compilation. It also broke subtitle rendering in
vo_rpi, which reuses some of the OpenGL code.
Line continuation was finally added in GLES 3.0, which is perhaps the
reason why ANGLE accepted it.
When using --start with timeline/ordered chapters, then the
timeline_switch_to_time() function will look at playback_initialized
whether to rselect the currently selected streams on the demuxer level.
So we need to set this field to true at an earlier stage during
initialization, and in particular before the code for --start is called.
Apparently it's only wine where the qpc_position returned by
IAudioClock_GetPosition can be overflowed. So actually do the rescaling
correctly, but throw away the result if it looks unreasonable.
this fixes a regression in 5afa68835a
In particular, the player inserts af_scaletempo by default now when
changing speed, so the suggestion to add this was missleading and
counter-productive.
Slightly change how it is decided when a new packet should be read.
Switch to demux_read_packet_async(), and let the player "wait properly"
until required subtitle packets arrive, instead of blocking everything.
Move distinguishing the cases of passive and active reading into the
demuxer, where it belongs.
Just so I can remove a few lines from dec_sub.c.
This is slightly inelegant, as the whole subtitle file has to be read
into memory, converted at once in memory, and then provided to
libavformat in an awkward way by creating a memory stream instead of
using demuxer->stream. It also won't be possible to force the charset on
subtitles in binary container formats - but this wasn't exposed before,
and we just hope this won't be ever needed. (One motivation was fixing
broken files with non-UTF8 muxed.) It also won't be possible to change
the charset on the fly, but this was not exposed either.
Just simplify by removing parts not needed anymore. This includes
merging dec_sub allocation and initialization (since things making
initialization complicated were removed), or format support queries (it
simply tries to create a decoder, and if that fails, tries the next
one).
So that the video FPs is not required at initialization, and can be set
later.
(As for whether this MicroDVD crap is worth the trouble to handle it
"correctly": MicroDVD files are unfortunately still around, and in at
least one case using the video FPS seemed to help indeed.)
Always preroll by default if the cue (index) information indicates
overlapping subtitles.
Increase the amount of maximum data it will skip to get such subtitles
to 10 seconds. Since the index information can reliably tell whether
reading earlier is needed, the maximum should be rarely actually used,
thus we can set it high. On the other hand, the "old" prerolling
mechanism always has to skip the maximum amount of data; thus the method
using the index gets its own option to control the maximum amount of
data to skip.
(As more and more files With newer mkvtoolnix versions are muxed, and
with this new and hopefully sane default established, these options can
probably be removed in the future.)
Untested, but should be fine. Broken by commit 0a0bb905.
Also fix the include statement in context_rpi.c, which caused another
compilation failure. Also untested. (Because I'm lazy.)
Fixes#2638.
Keeping ASS_Renderers around for a potentially large number of subtitle
tracks could lead to excessive memory usage, especially since the libass
cache is broken (caches even unneeded data), and might consume up to
~500MB of memory for no reason.
This includes the case of switching ordered chapter boundaries. It will
now be recreated on each timeline part switch. This shouldn't be much of
a problem with modern libass. (Older libass versions use fontconfig for
memory fonts, and will be very slow to reinitialize memory fonts.)
Since commit 6d9cb893, subtitle state doesn't survive timeline switches
(ordered chapters etc.). So there is no point in caching the state per
sh_stream anymore (which would be required to deal with multiple
segments). Move the cache to struct track.
(Whether it's worth caching the subtitle state just for the situation
when subtitle tracks get reselected is questionable. But for now, it's
nice to have the subtitles immediately show up when reselecting a
subtitle.)
For files with only 1 chapter, the "cycle" command was ignored. Reenable
it, but don't let it terminate playback of the file.
For the full story, see #2550.
Fixes#2550.
OK, this made the --sub-paths and --audio-file-paths synonyms, which is
not what we wanted. Actually restrict the type of file loaded as well.
Really fixes#2632.
Requested. It works like --sub-paths. This will also load audio files
from a "audio" sub directory in the config file (because the same code
as for subtitles is used, and it also had such a feature).
Fixes#2632.
When crossing timeline boundaries (such as switching to a new segment or
chapter with ordered chapters), clear the internal text subtitle list.
This breaks the sub-seek command, but is otherwise not too harmful.
Fixes Sub-OC-test-final7.mkv. (The internal text subtitle list is
basically a cache to make subtitles show up at the right time when
seeking back.)
I suspect this was caused by 76fcef61. The sample file times subtitles
slightly before the video frame when it should show up. This is to avoid
problems with subtitles showing up a frame later than intended. It also
means that a subtitle which is supposed to show up on the start of a
timeline part boundary actually might first be shown in a different
part. Since we now manipulate the packet timestamps, instead of
manipulating timestamps after the subtitle decoder, this means this
subtitle event would have 2 timestamps, which our code of course does
not handle.
If the two parts come one after another, this would actually work (since
the subtitle would have the same timestamps in the old and new part),
but it breaks if the new part (which follows the old part in the
physical file) is has a completely different start time in the timeline.
Essentially, the trick used to time subtitles correctly is incompatible
with the way we cache subtitles (to make them survive seeks).
The simple solution is just clearing the cached subtitles when crossing
chapter boundaries.
See #2609:
"When eof is reached it would be shown on the OSD and in the console.
Next try seeking to the middle. Seeking to the middle of the file will
only result in the OSD message being updated. Lua seems to fail to
observe the change in the property until the video is unpaused."
Merge blend_src8_alpha and blend_src16_alpha into blend_src_alpha, and
the same for blend_const_alpha.
One thing that changes is that the vertical loop is now shared for both
code paths.
I think this is slightly easier to read, and it's a bit shorter as well.
This removes the need to define IMGFMT_GBRAP, which fixes compilation
with the current Libav release.
This also makes it automatically pick up a GBRP format with the same bit
width. (Unfortunately, it seems libswscale does not support conversion
to AV_PIX_FMT_GBRAP16, so our code falls back to 8 bit, removing
precision for video covered by subtitles in cases this code is used.)
Also, when the source video is e.g. 10 bit YUV, upsample to 16 bit.
Whether this is good or bad, it fixes behavior with alpha. Although I'm
not sure if the alpha range is really correct ([0,2^16-1] vs.
[0,255*256]). Keep in mind that libswscale doesn't even agree with the
way we do it.
This actually treats destination alpha correctly, and gives much better
results than before. I don't know if this is perfectly correct yet,
though. Slight difference with vo_opengl behavior suggests it might not
be.
Note that this does not affect VOs with true alpha support. vo_opengl
does not use this code at all, and does the alpha calculations in OpenGL
instead.
Commit 1a6f3c56 added a fallback for the case when C11 TLS was not
available, but GCC TLS was. But it forget to enable VAAPI EGL interop in
the build system in this case.
Just remove the build system check. Should someone find a compiler that
works on Linux and does not support GCC extensions or C11, it will still
compile and just fail to init at runtime.
Actually fixes#2631 (hopefully).
The demuxer infrastructure was originally single-threaded. To make it
suitable for multithreading (specifically, demuxing and decoding on
separate threads), some sort of tripple-buffering was introduced. There
are separate "struct demuxer" allocations. The demuxer thread sets the
state on d_thread. If anything changes, the state is copied to d_buffer
(the copy is protected by a lock), and the decoder thread is notified.
Then the decoder thread copies the state from d_buffer to d_user (again
while holding a lock). This avoids the need for locking in the
demuxer/decoder code itself (only demux.c needs an internal, "invisible"
lock.)
Remove the streams/num_streams fields from this tripple-buffering
schema. Move them to the internal struct, and protect them with the
internal lock. Use accessors for read access outside of demux.c.
Other than replacing all field accesses with accessors, this separates
allocating and adding sh_streams. This is needed to avoid race
conditions. Before this change, this was awkwardly handled by first
initializing the sh_stream, and then sending a stream change event. Now
the stream is allocated, then initialized, and then declared as
immutable and added (at which point it becomes visible to the decoder
thread immediately).
This change is useful for PR #2626. And eventually, we should probably
get entirely of the tripple buffering, and this makes a nice first step.
The "script-binding" command is used by the Lua scripting wrapper to
register key bindings on the fly. It's also the only way to get fine-
grained information about key events (such as separate key up/down
events). This information is sent via a "key-binding" message when the
state of a key changes.
Extend it to send name of the mapped key itself. Previously, it was
assumed that the user just uses an unique identifier for the binding's
name, so it wasn't needed. With this change, a user can map exactly the
same command to multiple keys, which is useful especially with the next
commit.
Part of #2612.
gcc 4.8 does not support C11 thread local storage. This is a bit
annoying, so add a hack to use the gcc specific __thread extension if
C11 TLS is not available.
(This is used for the extremely silly mpv-internal way hwdec modules
access some platform specific handles. Disabling it simply made
hwdec_vaegl.c always fail initialization.)
Fixes#2631.
Add a "blend-tiles" choice to the "alpha" sub-option. This is pretty
simplistic and uses the GL raster position to derive the tiles. A weird
consequence is that using --vo=opengl and --vo=opengl-hq gives different
scaling behavior (screenspace pixel size vs. source video pixel size
16x16 tiles), but it seems we don't have easy access to the original
texture coordinates. Using the rasterpos is probably simpler.
Make this option the default.
Github will display a link to it when a user wants to create an issue or
pull request.
Also make some minor adjustments to DOCS/contribute.md, which is
developer oriented, and for which I see no reason to merge it with
the new file.
window-scale is now mapped to Alt+0 etc. by default (although these
bindings just use "set", not "cycle-values").
colormatrix can't be cycled anymore (would require using vf_format).
Make sure that subtraction of performance counters is done correctly.
Follow the *exact* instructions for converting performance counter to something
comparable to the QPCposition returned by IAudioClient::GetPosition
https://msdn.microsoft.com/en-us/library/windows/desktop/dd370889%28v=vs.85%29.aspx
Also make sure that subtraction of unsigned integers is stored into a signed
integer to avoid nastiness. Also be more careful about overflow in the
conversion of the device position into number of samples.
Avoid casting mp_time_us() to a double, and use llrint to convert the
double precision delay_us back to integer for ao_read_data.
Finally, actually check the return value of ao_read_data and add a verbose
message if it is not the expected value. Unfortunately,
there is no way to tell WASAPI when this happens since the frame_count in
ReleaseBuffer must match GetBuffer.
This is for the sake of command.c and the "deinterlace" option/property.
Instead of forcing certain "better" defaults when inserting yadif,
change the actual "yadif" defaults.
I pondered not changing vf_yadif, and instead adding a trivial "yadif-
auto" wrapper filter, which would merely have different defaults. But
thinking about it, it doesn't make any sense for "deinterlace" to have
different defaults from vf_yadif, with vf_yadif having the "worse"
defaults. If someone wants the old behavior, the old behavior can be
forced in a backward and forward compatible way by setting the
suboptions.
Fixes#2539 (kind of).
When libuchardet returns an empty string, it can be either ASCII, UTF-8,
or an unknown encoding. Try to distinguish it from the unknown case by
checking for UTF-8. This avoids an annoying message, and avoids
unnecessary processing (we convert invalid UTF-8 sequences to latin1 to
workaround libavcodec's braindead UTF-8 check).
Commit 127da161 was not properly tested either - it did nothing, and
just made it use the video bitstream aspect ratio determined by
libavformat (which isn't always the correct one).
long is 64 bits on x86_64 on Linux, which means the check for the corner
case of computing the depth mask is wrong.
Also, X11 compositors seem to expect premultiplied alpha.
Do not try and set/get master volume in exclusive if there is no
hardware support. This would just uselessly change the master slider,
but have no effect on the actual volume.
Furthermore if getting hardware volume support information fails, then assume
it has none.
It was complicated and not even very intuitive to the user.
If you are controlling the master volume, you just have to be
prepared to deal with the consequences.
Both mpv and ffmpeg have their own internal pthreads wrappers. The mpv
one has been recently enabled by default as well. (It didn't work on XP,
but we dropped XP support.)
libwaio was added due to the complete inability to cancel synchronous
I/O cleanly using the public Windows API in Windows XP. Even calling
TerminateThread on the thread performing I/O was a bad solution, because
the TerminateThread function in XP would leak the thread's stack.
In Vista and up, however, this is no longer a problem. CancelIoEx can
cancel synchronous I/O running on other threads, allowing the thread to
exit cleanly, so replace libwaio usage with native Vista API functions.
It should be noted that this change also removes the hack added in
8a27025 for preventing a deadlock that only seemed to happen in Windows
XP. KB2009703 says that Vista and up are not affected by this, due to a
change in the implementation of GetFileType, so the hack should not be
needed anymore.
This sets the minimum supported Windows version to Windows Vista. The
subsystem version also affects some Windows API functions, including
GetSystemMetrics(SM_CXPADDEDBORDER).
CSIDLs have been deprecated in Windows Vista and are not recommended for
use in new code. They have been replaced with Known Folder IDs, which
are pretty much the same thing, except they use GUIDs.
Converted subtitles use a different method to avoid adding repeated
packets as duplicate subtitle events. The state for this mechanism must
be cleared as well if --sub-clear-on-seek is used.
MPlayer traditionally always used the display aspect ratio, e.g. 16:9,
while FFmpeg uses the sample (aka pixel) aspect ratio.
Both have a bunch of advantages and disadvantages. Actually, it seems
using sample aspect ratio is generally nicer. The main reason for the
change is making mpv closer to how FFmpeg works in order to make life
easier. It's also nice that everything uses integer fractions instead
of floats now (except --video-aspect option/property).
Note that there is at least 1 user-visible change: vf_dsize now does
not set the display size, only the display aspect ratio. This is
because the image_params d_w/d_h fields did not just set the display
aspect, but also the size (except in encoding mode).
Since alpha isn't pulled through the colormatrix (maybe it should?), we
reject alpha formats with odd sizes, such as yuva444p10.
But the awful tex_mul path in vo_opengl does this anyway (at some points
even explicitly), which means there will be a subtle difference in
handling of 16 bit yuv alpha formats. Make it consistent and always
apply the range adjustment to the alpha component. This also means odd
sizes like 10 bit are supported now.
This assumes alpha uses the same "shifted" range as the yuv color
channels for depths larger than 8 bit. I'm not sure whether this is
actually the case.
Now common.c only contains the code for the function loader, while
context.c contains the backend loader/dispatcher.
Not calling it "backend.c", because the central struct is called
MPGLContext.
This is used for dithering, although I'm not aware of anyone who got
higher than 8 bit depth support to work on Linux.
Also put this into egl_helpers.c. Since EGL is pseudo-portable at best I
have no hope that the EGL context creation code in all the backends can
be fully shared. But some self-contained functionality can definitely be
shared.
Store the determined framebuffer depth in struct GL instead of
MPGLContext. This means gl_video_set_output_depth() can be removed, and
also justifies adding new fields describing framebuffer/backend
properties to struct GL instead of having to add more functions just to
shovel the information around.
Keep in mind that mpgl_load_functions() will wipe struct GL, so the
new fields must be set before calling it.
Although the source file is named w32.c, the backend name was "win"
until recently. It was accidentally changed to "w32"; fix it.
Fixes#2608 (the manual is correct).
This actually alows to playback alternating videos with mpv.
Tested with actual file found in wild remuxed to mkv and changed props
with following command:
mkvpropedit /tmp/o.mkv --edit track:1 -s stereo-mode=13
Signed-off-by: Paul B Mahol <onemda@gmail.com>
Apparently, this was replaced by the SD_CTRL_SET_VIDEO_PARAMS set
dimensions. But I can't find out when this happened - possibly, these
fields were never used by sd_lavc.c, and only by the (long removed)
MPlayer dvdsub decoder.
The previous commit turned sd_lavc_conv from a sd_driver to
free-standing functions. Do the rename to reflect this change
separately to avoid confusing git's content tracking. (Or did
git solve this, making separating renames and content changes
unnecessary?)
It was stupid. The only thing that still effectively used it was
sd_lavc_conv - all other "filters" were the subtitle decoder/renderers
for text (sd_ass) and bitmap (sd_lavc) subtitles.
While having a subtitle filter chain was interesting (and actually
worked in almost the same way as the audio/video ones), I didn't
manage to use it in a meaningful way, and I couldn't e.g. factor
secondary features like fixing subtitle timing into filters.
Refactor the shit and drop unneeded things as it goes.
This affects non-ASS text subtitles (those which go through libavcodec's
subtitle converter), which are muxed with video/audio. (Typically srt
subs in mkv.)
The problem is that seeking in the file can send a subtitle packet to
the decoder multiple times. These packets are interlaved with video,
and thus can't be all read when opening the file. Rather, subtitle
packets can essentially be randomly skipped or repeated (by seeking).
Until recently, this was solved by scanning the libass event list for
duplicates. Then our builtin srt-to-ass converter was removed, and
the problem was handled by fully clearing the subtitle list on each
seek.
This resulted in sub-seek not working properly for this type of file.
Since the subtitle list was cleared on seek, it was not possible to
do e.g. sub-seeks to subtitles before the current playback position.
Fix this by not clearing the list, and intead explicitly rejecting
duplicate packets. We use the packet file position was unique ID for
subtitles; this is confirmed working for most file formats (although
it is slightly risky - new demuxers may not necessarily set the file
position to something unique, or at all).
The list of seen packets is sorted, and the lookup uses binary search.
This is to avoid quadratic complexity when subtitles are added in
bulks, such as when opening a text subtitle file.
In some places, the code has to be adjusted to pass through the packet
file position correctly.
This can happen if the file references a track, but does not specify
an INDEX 01 for it. This would cause mpv to just segfault due to
dereferencing the null pointer as a string.
A file causing this was observed in the wild by
ExactAudioCopy v0.99pb4 for a disk that contained a data track at the
end.
With the FFmpeg subtitle decoder used for _all_ non-ASS text subtitle
format, this code is simply unused now.
Ironically, the FFmpeg subtitle decoder does not handle things correctly
in a bunch of cases. Should it turn out they actually matter, they will
have to hack back.
The extend_event one is a candidate, although even though there were
allegedly files which need it, I couldn't get samples from the user who
originally reported such files. As such, extend_event was only confirmed
to handle trailing events with no (endless) duration like with MicroDVD
and LRC, but FFmpeg "fudges" these anyway, so no special handling is
needed.
This code also had logic to handle seeking with muxed srt subtitles,
which made the sub-seek command work. But this has been broken before
this commit already. Currently, seeking with muxed srt subs will clear
all subtitles, as the broken FFmpeg ASS format output by the libavcodec
subtitle converters does not check for duplicates. Since the subtitles
are all cleared, ass_step_sub() can not work properly and sub-seek can
not seek to already seen subtitles.
Slightly simpler, and removes the need to pre-read all subtitle packets.
This still does the subtitle charset conversion on the packet level
(instead converting when parsing the file), so in theory this still
could provide a way to change the charset at runtime. But maybe even
this should be removed, as FFmpeg is somewhat likely to get its own
charset detection and conversion mechanism in the future. (Would have
to keep the subtitle file in memory to allow changing the charset on
the fly, I guess.)
The FFmpeg subtitle converter does the same. There used to be some
deficiencies in FFmpeg's code, but it seems at least some of them have
been fixed. There also used to be the timestamp issue (see previous
commit messages), but this doesn't matter anymore. So no reason to
keep this code - get rid of it.
This can be dropped for the same reasons as in the previous commits. It
removes MicroDVD conversion support on Libav, although MicroDVD files
couldn't be read in the first place ever since demux_subreader.c was
removed.
This restored timestamps when demuxing srt subtitles in Libav, which
was important for avoiding slightly overlapping subtitles. Since the
way this works was changed, there is no real reason to maintain proper
timestamps anymore on this level - this can be dropped without issues.
When a Direct3D 9Ex device fails to reset, it gets put into the lost
state, so set the lost_device flag and don't attempt to present until
the device moves out of that state. Failure to recreate the size-
dependent objects should set lost_device as well, since we shouldn't try
to present in that state.
Also, it looks like I was too eager to remove code that sets priv
members to NULL and I accidentally removed some that was needed.
Direct3D doesn't like 0-sized swapchain dimensions, even when those
dimensions are automatically set. Manually set them to a size that isn't
zero instead.
Why not.
Also, instead of disabling hue/saturation for RGB, just don't apply
them. (They don't make sense for conversion matrixes other than YUV, but
I can't be bothered to keep the fine-grained disabling of UI controls
either.)
A manually added af_volume could lead to muted audio when switching to a
new file. af_volume keeps the last volume set by AF_CONTROL_SET_VOLUME
to return it with AF_CONTROL_GET_VOLUME, but the initial value is 0. So
the mixer volume was forced to 0 when unintializing the filter chain and
reading back the previously set volume.
WGL_NV_DX_interop is widely supported by Nvidia and AMD drivers. It
allows a texture to be shared between Direct3D and WGL, so that
rendering can be done with WGL and presentation can be done with
Direct3D. This should allow us to work around some persistent WGL
issues, such as dropped frames with some driver/OS combos, drivers that
buffer frames to increase performance at the cost of latency, and the
inability to disable exclusive fullscreen mode when using WGL to render
to a fullscreen window.
The addition of a DX_interop backend might also enable some cool
Direct3D-specific enhancements in the future, such as using the
GetPresentStatistics API to get accurate frame presentation timestamps.
Note that due to a driver bug, this backend is currently broken on
Intel. It will appear to work as long as the window is not resized too
often, but after a few changes of size it will be unable to share the
newly created renderbuffer with GL. See:
https://software.intel.com/en-us/forums/graphics-driver-bug-reporting/topic/562051
All of these are supported by FFmpeg now. It was disabled by default
too (with FFmpeg).
If compiled against Libav, mpv will lose the ability to read some
subtitle formats (but the most important ones, srt and ass, still should
work).
This is only for specific Hauppage cards. According to the comments in
who is actively using this feature. Get it out of the way.
Anyone who still wants to use this should complain. Keeping this code
would not cause terribly much additional work, and it could be restored
again. (But not if the request comes months later.)
With default setting, the matrix for fruit dithering requires 12 bits
precision (values from 0/4096 to 4095/4096). But 16-bit float
provides only 10 bits. In addition, when `dither-size-fruit=8` is
set, 16 bits are required from the texture format.
Fix this by attempting to use 16 bit integer texture first. This is
still not precise, but should be better than using a half float.
This has no reason to be there. Put the functionality into another
function instead. While we're at it, also adjust for possible accuracy
issues with high bit depth YUV (matters for rendering subtitles into
screenshots only).
The recent LUT adjustment changes broke interpolation.
The concatenation of the shader stages is a bit messy, and it seems like
sampler_prelude is not a good place to add this macro. Always add the
macro to every shader instead. (While this doesn't seem too elegant,
this isn't too inelegant either, and goes these problems out of the
way.)
The computation of the tex_mul variable was broken in multiple ways.
This variable is used e.g. by debanding for moving expansion of 10 bit
fixed-point input to normalized range to another stage of processing.
One obvious bug was that the rgb555 pixel format was broken. This format
has component_bits=5, but obviously it's already sampled in normalized
range, and does not need expansion. The tex_mul-free code path avoids
this by not using the colormatrix. (The code was originally designed to
work around dealing with the generally complicated pixel formats by only
using the colormatrix in the YUV case.)
Another possible bug was with 10 bit input. It expanded the input by
bringing the [0,2^10) range to [0,1], and then treating the expanded
input as 16 bit input. I didn't bother to check what this actually
computed, but it's somewhat likely it was wrong anyway. Now it uses
mp_get_csp_mul(), and disables expansion when computing the YUV matrix.
It turns out that with accurate lookup we can decrease the
default size of texture now. Do it to compensate the performance
loss introduced by the LUT_POS macro.
Define a macro to correct the coordinate for lookup texture. Cache
the corrected coordinate for 1D filter and use mix() to minimize the
performance impact.
It always was a weird artifact - VOCTRLs are meant _not_ to require
special handling in the code that passes them through (like in vo.c).
Removing it is also interesting to further reduce the dependency of
backends on struct vo. Just get rid of it.
Removing it is somewhat inconvenient, because in many situations the UI
window is created after the first VOCTRL_UPDATE_WINDOW_TITLE. This means
these backends have to store it in a new field in their own context.
Fixes a reported sample, that has a sign interrupted by a few frames
(for which --sub-fix-timing would remove the wanted gap).
The list of tags in has_overrides() is taken from libass. It has a
similar function (which even checks whether the tag are within the { }
delimiters). Unfortunately, this function is not public, so we just have
a simpler one which does roughly the same. It doesn't matter that this
function sometimes returns false positives.
The awkward "preprocess" step of putting the subtitles through single
filters before doing something else was made unnecessary by the recent
changes.
(Fun fact: I originally planned to move these extra things, like fixing
subtitle gaps/overlaps, to filters - but this would suffer from various
complications, and moving them to the renderers seems much simpler.)
I feel like it's better there. Note that there is no reduced
functionality, as bitmaps subs (i.e. not handled by sd_ass.c) were never
fully read on init, and thus never went through sub_read_all_packets().
On the other hand, this might lead to confusion, as --sub-fps etc. will
now also affect muxed subtitles (which makes not much sense).
Instead of messing with the subtitle packet timestamps, do it on output.
We work on the libass event list. If there is an unwanted gap or
overlap, we render the timestamp at another position where there is no
gap or overlap.
This is somewhat more robust, and even works with demuxed subs (to some
degree - depends whether the subs are prefected soon enough).
It's active even for native ASS subs. I wonder if this is a problem with
extended type setting. If it is, the heuristic that tries to avoid
interrupting such cases has to be improved.
While it probably would be ideal to do this after the subtitle decoder,
certain aspects are at least currently handled better in this place.
Image subtitles often use a "signaling" packet to set the end time of
the previous subtitle. As far as the libavcodec API is concerned, such
packets decode to empty AVSubtitles. Discard these after the end time of
the previous subtitle has been set.
Keep track of the per-subtitle end time better. This is for the sake of
improving sub_step/sub_seek. Without this, it would seek to the sub
before the previous sub, if the current sub has ended displaying.
Works roughly the same as the one in sd_ass for text subtitles. While
sub_step is very uninteresting, it comes for free with the support for
sub_seek.
The implementation is taken from ass_step_sub() from libass, with some
modifications
Until now, feeding packets to the decoder in advance was done for text
subtitles only. This was possible because libass buffers all subtitle
data anyway (in ASS_Track). sd_lavc, responsible for bitmap subs, does
not do this. But it can buffer a small number of subtitle frames ahead.
Enable this.
Repurpose the sub_accept_packets_in_advance(). Instead of "can take all
packets" it means "can take 1 packet" now. (The old meaning is still
needed locally in dec_sub.c; keep it there.) It asks the decoder whether
there is place for at least 1 subtitle packet. sd_lavc implements it and
returns true if its internal fixed-size subtitle queue still has a free
slot. (The implementation of this in dec_sub.c isn't entirely clean.
For one, decode_chain() ignores this mechanism, so it's implied that
bitmap subtitles do not use the subtitle filter chain in any advanced
way.)
Also fix 2 bugs in the sd_lavc queue handling. Subtitles must be checked
in reverse, because the first entry will often have endpts==NOPTS, which
would always match. alloc_sub() must cycle the queue buffer, because it
reuses memory allocations (like sub.imgs) by design.
If reinit after a fallback from hardware fails, this field can be NULL.
The check in control() was broken due to a typo (found by Coverity), and
decode() lacked the check entirely.
If the sampling point is placed diagonally, the radius difference
could be as large as sqrt(2.0). And a loosened check with (radius - 1)
would potentially include pixels out of the range.
Fix the check to handle those corner case properly to avoid
unnecessary texture lookup and improve the performance a bit.
reopen_console_handle() was never properly tested because mpv overrides
printf in most source files. Turns out that when there's no console on
startup, the CRT sets the fds of stdout and stderr to -2, so the old
method of using dup2 to manipulate these fds didn't work. As far as I
can tell, the only way to give stdout and stderr valid fds is to use
freopen, so this uses freopen to set them both to the console output.
This also uses dup2 to change STDOUT_FILENO and STDERR_FILENO, so low-
level functions like isatty still work.
Note that this means fileno(stdout) != STDOUT_FILENO. I don't think this
will cause any problems.
This should fix MPV_LEAK_REPORT on the Windows console.
Helps with files that have occasional broken timestamps. For larger
discontinuities, e.g. caused by actual timestamp resets, we still want
to realign audio.
(I guess in general, this should be removed and replaced by a more
general resync-on-desync logic, but not now.)
Sometimes QOpenGLWidget may be redirecting it's output to a framebuffer
object rather than the frontbuffer, in which case the current thread's
context render fbo is different from the widget's. Use the widget's
desired fbo instead.
There are claims that nnedi3.c doesn't constitute its own new
implementation, but is derived from existing HLSL or OpenCL shaders
distributed under the LGPLv3 license.
Until these are resolved, do the "correct" thing and require
--enable-gpl3 to build nnedi.
This makes no sense, because the client is obligated to react to this
event.
This also happens to fix a deadlock with JSON IPC clients sending
"disable_event all", because MPV_EVENT_SHUTDOWN was used to stop the
thread driving the socket connection (fixes#2558).
Approximately reverts commit 3ccac74d. This failed with some avi files,
which do pseudo-VFR by sending packets with empty frames (or repeat
frames, depending on point of view). Specifically, these packets are not
0 bytes, so they don't get skipped by libavformat, as with the usual VFR
avi hack. Instead, the packet contains a VOP with vop_coded=0, so
libavcodec will just return no frame. We could probably distinguish such
skipped frames and delayed frames by explicitly measuring the codec
delay by counting how long it takes to get the very first frame (and
then treat skips as explicit drops), but we may as well simply reinstate
the old code.
To appease to at least one semi-broken case, do not enable this logic on
the RPI, as the FFmpeg MMAL wrapper has arbitrary buffering (and MMAL
itself is asynchronous).
I guess gl_video->global was originally meant to be optional, but now it
crashes in some newer code with vo_opengl_cb, which tries to init it
with this field set to NULL (because normally it's not needed).
Probably fixes#2542.
Requested. Don't overwrite permanent OSD text set with e.g. --osd-msg1.
Instead, append the OSD message to it (on the next line).
Note that with --osd-msg1, seeking will still overwrite the OSD with the
playback status for a while. If you do not want this, use --osd-msg3
--osd-level=3 instead.
Don't use the average FPS if there are likely skipped vsyncs. Note that
we don't use the normal skip detection, as it is unreliable if the real
and assumed display FPS differ too much. The normal skip detection is
still in place as it's more reliable in the case when vsync jitters
much, but the display FPS is relatively exact.
Further improvement over commit 41f2c653.
Commit 2b07d3eb merged progbar and OSD text renderer into one ASS_Track,
but it confused the styles. Specifically, if both progbar and OSD are
visible, the create_ass_track() call made by the progbar code will reset
the style adjusted by the OSD text code.
Change create_ass_track() not to add any styles. Instead let the caller
manage the styles. They are now referenced by name, and lazily added if
they don't exist yet. This is also much cleaner.
Reduces memory usage and startup times. The implementation is a bit
weird, because both OSD parts have conflicting requirements on the used
ASS styles.
At least I hope so.
Deriving the duration from the pts was not really correct. It doesn't
include speed adjustments, and becomes completely wrong of the user e.g.
changes the playback speed by a huge amount. Pass through the accurate
duration value by adding a new vo_frame field.
The value for vsync_offset was not correct either. We don't need the
error for the next frame, but the error for the current one. This wasn't
noticed because it makes no difference in symmetric cases, like 24 fps
on 60 Hz.
I'm still not entirely confident in the correctness of this, but it sure
is an improvement.
Also, remove the MP_STATS() calls - they're not really useful to debug
anything anymore.
This was just converting back and forth between int64_t/microseconds and
double/seconds. Remove this stupidity. The pts/duration fields are still
in microseconds, but they have no meaning in the display-sync case (also
drop printing the pts field from opengl/video.c - it's always 0).
If we switched away from the system FPS, we were remaining in this mode
ssentially forever. There's no reason to do so; switch back if the
estimated FPS gets worse again. Improvement over commit 41f2c653.
Commit 12eb8b2d accidentally disabled framedropping in the audio timing
case. It tried to replace the last_flip field with the prev_vsync one,
which didn't work because prev_sync is reset to 0 if the timing code is
used. Fix it by always setting it properly. This field must (or should)
be reinitialized to something sensible when switching to display sync
timing mode; since prev_vsync is not reset anymore, the check when to
reinitialize this field has to be adjusted as well.
It's a bit weird that update_vsync_timing_after_swap() now does some
minor work for timing mode too, but I guess it's ok, if only to avoid
additional fields and timer calls.
If there were many AO drivers without device selection, this added a
"Default" entry for each AO. These entries were not distinguishable, as
the device list feature is meant not to require to display the "raw"
device name in GUIs.
Disambiguate them by adding the driver name. If the AO is the first, the
name will remain just "Default". (The condition checks "num > 1",
because the very first entry is the dummy for AO autoselection.)
Instead of periodically trying to enable it again. There are two cases
that can happen:
1. A random discontinuity messed everything up,
2. Things are just broken and will desync all the time
Until now, it tried to deal with case 1 - but maybe this is really rare,
and we don't really need to care about it. On the other hand, case 2 is
kind of hard to diagnose if the user doesn't use the terminal.
Seeking will reenable display-sync, so you can fix playback if case 1
happens, but still get predictable behavior in case 2.
If the system-reported display FPS (returned by the VO backends, or
forced with --display-fps) is too imprecise (deviating frame duration by
more than 1%). This works if the display FPS is off by almost 1 (typical
for old/bad/broken OS APIs). Actually it even works if the FPs is
completely wrong.
Is it a good idea? Probably not. It might be better to only output a
warning message. But unless there are reports about it going terribly
wrong, I'll go with this for now.
Actually I'm not content with the detection added in commit 44376d2d. It
triggers too often if vsync is very jittery. It's easy to avoid this: we
add more samples to the detection by reusing the drift computation loop.
If there's a significant step in the drift, we consider it a drop.
This adds basic support for ICC profiles. Per-monitor profiles are
supported. WCS profiles are not supported, but there is an API for
converting WCS profiles to ICC, so they might be supported in future.
I'm just not sure if anyone actually uses them.
Reloading the ICC profile when it's changed in the control panel is also
not supported. This might be possible by using the WCS APIs and watching
the registry for changes, but there is no official API for it, and as
far as I can tell, no other Windows programs can do it.
Of course, only FFmpeg has av_clipd(), while Libav does not. (Nevermind
that it doesn't do much more than the mpv MPCLAMP() macro. Supposedly,
libavutil can provide optimized platform-specific versions for av_clip*,
but of course nothing actually does for av_clipf() or av_clipd().)
Return the estimated/ideal flip time to the timing logic (meaning
vo_get_delay() returns a smoothed out time). In addition to this add
some lame but working drift compensation. (Useful especially if the
display FPS is wrong by a factor such as 1.001.)
Also remove some older leftovers. The vsync_interval_approx and
last_flip fields are redundant or unneeded.
For the vo-delayed-frame-count property.
Slightly less dumb than the previous one (which was removed earlier),
but still pretty dumb. But this also seems to be relatively robust, even
with strong vsync jittering.
This logic was kind of questionable anyway, and --display-sync should
give much better results. (I would even go as far as saying that the
FPS-dependent framedrop code made things worse in some situations. Not
all, though.)
This is simply the average refresh rate. Including "bad" samples is
actually an advantage, because the property exists only for
informational purposes, and will reflect problems such as the driver
skipping a vsync.
Also export the standard deviation of the vsync frame duration
(normalized to the range 0-1) as vsync-jitter property.
libswresample doesn't do it - although it should, but the patch is stuck
in limbo.
Probably reduces problems with artifacts on downmixing in some cases.
The OSD takes up an entire fullscreen dispmanx layer. Although the GPU
should be able to handle it (possibly even without any disadvantages),
it'll still be useful for debugging performance issues.
This is a hack, but unfortunately the DwmGetCompositionTimingInfo
heuristic does not work in all cases (with multiple-monitors on Windows
8.1 and even with a single monitor in Windows 10.) See the comment in
mp_w32_is_in_exclusive_mode() for more details.
It should go without saying that if any better method of doing this
reveals itself, this hack should be dropped.
Remove known useless device entries from the --audio-device list (and
corresponding property). Do this because the list is supposed to be a
high level list of devices the user can select. ALSA does not provide
such a list (in an useable manner), and ao_alsa.c is still in the best
position to improve the situation somewhat.
The ALSA doxygen says:
IOID - input / output identification ("Input" or "Output"), NULL
means both
This bug was blatantly introduced with commit cf94fce4.
It doesn't have any real purpose anymore. Up until now, it was still
implemented by vo_wayland, but since we changed how the frame callbacks
work, even that appears to be pointless.
Originally, the plan was to somehow extend this mechanism to all
backends and to magically fix frame scheduling, but since we can't hope
for proper mechanisms even on wayland, this idea looks way less
interesting.
10 bit HEVC would require DXVA2_ModeHEVC_VLD_Main10, and most a
different surface type (judging by lavfsplitter source code, both
P010 and P016 would work). Since I'm unable to test this stuff,
exclude 10 bit for now.
See #2516.
If a second subtitle is shown, it should be forced to display on top
of the screen. This was working only if --no-ass was passed, because
otherwise the subtitle was rendered normally (i.e. usually on the
bottom).
This is actually all bullshit. There are many factors that can ruin the
video playback experience (and most outside of our or the user's
control).
One thing that makes sense is that this declares incompatibility with
Windows XP (fixes#2473).
The D3D9 backend does not support GLES 3, which makes it pretty useless.
But it still might be a legitimate replacement of vo_direct3d.c on
Windows 7 machines.
Note that we could just use:
eglGetDisplay(EGL_D3D11_ELSE_D3D9_DISPLAY_ANGLE)
But for now I'll leave the old code. Maybe this can exclude use of
software rendering backends (EGL_PLATFORM_ANGLE_DEVICE_TYPE_WARP_ANGLE).
Since I'm not sure, I won't touch it.
Running mpv with default config will now pick up ANGLE by default. Since
some think ANGLE is still not good enough for hq features, extend the
"es" option to reject GLES backends, and add to to the opengl-hq preset.
One consequence is that mpv will by default use libswscale to convert
10 bit video to 8 bit, before it reaches the VO.
I decided that I actually can't stand how vo_opengl unnecessarily puts
the video through 3 shader stages (instead of 1). Thus, what was meant
to be a fallback for weak OpenGL implementations, the dumb-mode, now
becomes default if the user settings allow it.
The code required to check for the settings isn't so wild, so I guess
it's manageable. I still hope that one day, our rendering logic can
generate ideal shader stages for this case too.
Note that in theory, dumb-mode could be reenabled at runtime due to a
color management 3D LUT being set, so a separate dumb_mode field is
required. The dumb-mode option can't just be overwritten.
Unfortunately, color management can still not work, because no GLES
version specified so far support fixed-point 16 bit textures. Maybe
we could use integer textures, but these don't support filtering.
Using float textures would be another possibility.
Polar scalers use 1D textures, because they're slightly faster on some
GPUs than 2D textures. But 2D textures work too, so add support for
them.
Allows using these scalers with ANGLE.
Just like commit f9a2fc59. There are probably some more such cases.
The vec2 constructor calls are probably fine, but don't bother with
confusing inconsistencies.
While desktop GL's glTexImage2D() essentially accepts anything, GLES is
much stricter. The combination of allowed formats/types/internal formats
is exactly specified. The GLES 3.0.4 specification lists them in
table 3.2. (The ANGLE API validation code references this table.)
The table could probably be extended into a general declarative table
about GL formats covering other uses, but this would be a big
non-trivial project, so don't bother and accept a minor degree
of duplication with other tables.
Note that the format and type do (or should) not matter here, because
no image data is transferred to the GPU.
We don't only need float textures for advanced scaling - we also need
them to be filterable with GL_LINEAR. On GLES, this is not supported
until GLES 3.1, but some implementation expose them with extensions.
This makes advanced scaling sort-of work for GLES 3.0 (on ANGLE). It's
still not very advisable, as 8 bits might not be enough to avoid
debanding. (Ironically, the debanding filter can be enabled, and does
not raise any GL errors - but probably doesn't do anything useful.)
Turns out glGetTexLevelParameter, which is missing in ANGLE, is a
GLES3.1 function. Removing it from the list of core GLES3 functions
makes ANGLE work in GLES3 mode.
Apparently, some audio drivers do not support the DTS subtype, but
passthrough works anyway if the AC3 subtype is set. Just retry with
AC3 if the proper format doesn't work. The audio device which
exposed this behavior reported itself as
"M601d-A3/A3R (Intel(R) Display Audio)".
xbmc/kodi even always passes DTS as AC3.
ANGLE is a GLES2 implementation for Windows that uses Direct3D 11 for
rendering, enabling vo_opengl to work on systems with poor OpenGL
drivers and bypassing some of the problems with native GL, such as VSync
in fullscreen mode.
Unfortunately, using GLES2 means that most of vo_opengl's advanced
features will not work, however ANGLE is under rapid development and
GLES3 support is supposed to be coming soon.
This is another regression of the recently added start time probing. If
a seek is executed after opening the file (but before reading any
packets), the first block is discarded instead of indexed. If there are
no other keyframes in the file, seeking will fail completely.
Fix it by seeking to the cluster start if there aren't any index entries
yet. This will read the skipped packet again.
Fixes#2498.
This was used with --no-sub-ass (aka --no-ass). This option (which is
not yet removed) strips all styling from the subtitles, and renders them
as plaintext only. For some reason, it originally seemed convenient to
reuse all the OSD text rendering code (osd_libass.c). While this was
indeed simple, it had a bad influence on the rest of the code. For
example, it had to decide whether to go through the OSD code path, or
the proper subtitle renderer in sd_ass.c.
Kill the OSD subtitle renderer. Reimplement --no-sub-ass and also
"secondary" subtitles in sd_ass.c. fill_plaintext() contains some rather
minor code duplication with osd_libass.c for setting up a dummy
ASS_Event and escaping the stripped text. Since sd_ass.c already has to
handle "normal" text subtitles, and has code for stripping ASS tags,
this remains all relatively simple.
Remove all the unnecessary crap from the rest of the code.
Use the demux_set_ts_offset() added in the previous commit to base each
timeline segment to use timestamps according to its relative position
within the overall timeline. As a consequence we don't need to care
about these timestamps anymore, and everything becomes simpler.
(Another minor but delicious nugget of sanity.)
Most of this is explained in the DOCS additions.
This gives us slightly more sanity, because there is less interaction
between the various parts. The goal is getting rid of the video_offset
entirely.
The simplification extends to the user API. In particular, we don't need
to fix missing parts in the API, such as the lack for a seek command
that seeks relatively to the start time. All these things are now
transparent.
(If someone really wants to know the real timestamps/start time, new
properties would have to be added.)
Something goes wrong somewhere. Don't bother, it's only needed for
compatibility with our absolute baseline (GL 2.1/GLES 2).
On the other hand, we can process nv12 formats just fine.
For the sake of vaapi interop, we want to use EGL, but on the other
hand, but because driver developers are full of shit, vdpau interop will
not work on EGL (even if the driver supports EGL). The latter happens
with both nvidia and AMD Mesa drivers.
Additionally, EGL vaapi interop support can apparently only detected at
runtime by actually using it. While hwdec_vaegl.c already does this, it
would require initializing libva on _every_ system, which will cause
libav to print an unpreventable bullshit message to the terminal.
Try to counter these huge loads of bullshit by adding more fucking
bullshit.
We want the following behavior:
- VO probed, backend probed: only accept non-sw, fail completely
otherwise
- VO forced, backend probed: use the first non-sw, or if none is found,
fall back to the first working sw backend
- VO probed, backend forced: (I don't care about this case)
- VO forced, backend forced: just use that backend
Also, on backend probe failure the vo->probed field was left in its old
state.
This adds support for the progress indicator taskbar extension
that was introduced with Windows 7 and Windows Server 2008 R2.
I don’t like this solution because it keeps its own state and
introduces another VOCTRL, but I couldn’t come up with anything
less messy.
closes#2399
In the display-sync, non-interpolation case, and if the display refresh
rate is higher than the video framerate, we duplicate display frames by
rendering exactly the same screen again. The redrawing is cached with a
FBO to speed up the repeat.
Use glBlitFramebuffer() instead of another shader pass. It should be
faster.
For some reason, post-process was run again on each display refresh.
Stop doing this, which should also be slightly faster. The only
disadvantage is that temporal dithering will be run only once per video
frame, but I can live with this.
One aspect is messy: clearing the background is done at the start on the
target framebuffer, so to avoid clearing twice and duplicating the code,
only copy the part of the framebuffer that contains the rendered video.
(Which also gets slightly messy - needs to compensate for coordinate
system flipping.)
Currently, vo.c will always continue to render the currently queued
frame, which sets last_flip, which in turn confuses vo_get_delay(),
which in turn will show a bogus A/V desync message on unpause. So just
reset it again on unpause.
I guess the removed code is an old leftover, and makes no sense anymore.
Should fix weird A/V diff dropouts when frames are being dropped with
display-sync.
If the player sends a frame with duration==0 to the VO, it can trivially
underrun. Don't panic, but keep the correct time.
Also, returning the absolute time from vo_get_next_frame_start_time()
just to turn it into a float with relative time was silly. Rename it and
make it return what the caller needs.
80ms allowable desync was a bit too much. It'd allow for a range of
160ms, which everyone can notice. It might also be a bother to apply
compensation resampling speed for that long.
We always let audio slowly desync until a threshold is reached, and then
pushed it back by applying a maximum compensation speed. Refine what
comes afterwards: instead of playing with the nominal video speed, use
the actual required audio speed for keeping sync as measured by the A/V
difference. (The "actual" speed is the ideal speed with A/V differences
added.)
Although this works in theory, it's somewhat questionable how much this
works in practice. The ideal time value is actually not exact, but is
the time at which the frame is scheduled (could be compensated by using
the time_left calculations in handle_display_sync_frame()). It doesn't
account for speed changes or catastrophic discontinuities. It uses only
10 past frames.
As long as it's within the desync tolerance, do not change the audio
speed at all for resampling. This reduces speed changes which might be
caused by jittering timestamps and similar cases.
(While in theory you could just not care and change speed every single
frame, I'm afraid that such changes could possibly cause audio
artifacts. So better just avoid it in the first place.)
This is very "illustrative", unlike the video-speed-correction
property, and thus useful. It can also be used to observe scheduling
errors, which are not detected by the core. (These happen due to
rounding errors; possibly not evne our fault, but coming from
files with rounded timestamps and so on.)
Instead of looking at the current frame duration for the intended
speedup, look at all past frames, and find a good average speed. This
ties in with not wanting to average _all_ frame durations, which
doesn't make sense in VFR situations.
This is currently done in the most naive way possible, but already sort
of works for VFR which switches between frame durations that are
integer multiples of a base rate. Certainly more improvements could
be made, such as trying to adjust directly on FPS changes, instead of
averaging everything, but for now this is not needed at all.
Helps somewhat with muxer-rounded timestamps.
There is some danger that this introduces a timestamp drift. But since
they are averaged values (unlike as when using an incorrect container
framerate hint), any potential drift shouldn't be too brutal, or
compensate itself soon. So I won't bother yet with comparing the results
with the real timestamp, unless we run into actual problems.
Of course we still prefer potentially real timestamps over the
approximated ones. But unless the timestamps match the container FPS,
we can't know whether they are (no, checking whether the they have
microsecond components would be cheating). Perhaps in future, we could
let the demuxer export the timebase - if the timebase is not 1000 (or
divisible by it), we know that millisecond-rounded timestamps won't
happen.
Get rid of get_past_frame_durations(), which was a bit too messy. Add
a past_frames array, which contains the same information in a more
reasonable way. This also means that we can get the exact current and
past frame durations without going through awful stuff. (The main
problem is that vo_pts_history contains future frames as well, which is
needed for frame backstepping etc., but gets in the way here.)
Also disable the automatic disabling of display-sync if the frame
duration changes, and extend the frame durations allowed for display
sync. To allow arbitrarily high durations, vo.c needs to be changed
to pause and potentially redraw OSD while showing a single frame, so
they're still limited.
In an attempt to deal with VFR, calculate the overall speed using the
average FPS. The frame scheduling itself does not use the average FPS,
but the duration of the current frame. This does not work too well,
but provides a good base for further improvements.
Where this commit actually helps a lot is dealing with rounded
timestamps, e.g. if the container framerate is wrong or unknown, or
if the muxer wrote incorrectly rounded timestamps. While the rounding
errors apparently can't be get rid of completely in the general case,
this is still much better than e.g. disabling display-sync completely
just because some frame durations go out of bounds.
We need a frame duration even on start, because the number of vsyncs
the frame is shown is predetermined. (vo_opengl actually makes use of
this property in certain cases.)
"Missed" implies the frame was dropped, but what really happens is that
the following frame will be shown later than intended (due to the
current frame skipping a vsync).
(As of this commit, this property is still inactive and always
returns 0. See git blame for details.)
Apparently Windows treats windows that use OpenGL, cover an entire
screen and have the WS_POPUP style set or are topmost windows as
exclusive fullscreen windows that bypass DWM and cannot be covered
by other windows.
This means we can’t use dwmflush in fullscreen mode, and it also
means that no other window can cover mpv, and it makes the screen
flicker when switching to fullscreen mode.
This can be avoided by not setting the WS_POPUP flag.
Users can still access the old behavior by enabling stay-on-top
(which IMO at least makes sense—now we just need to get dwmflush
autodetection right to avoid nasty surprises).
fixes#2177
This loaded external .ass files via libass. libavformat's .ass reader is
now good enough, so use that instead.
Apparently libavformat still doesn't support fonts embedded into text
.ass files, but support for this has been accidentally broken in mpv for
a while anyway. (And only 1 person complained.)
Just set the ratio directly by working around the intended semantics of
the API function. The silly rounding stuff we had isn't needed anymore
(and not entirely correct anyway).
Note that since the compensation is virtually active forever, we need to
reset if it's not needed. So always run this code to be sure to reset
it.
Also note that libswresample itself had a precision issue, until it
was fixed in FFmpeg commit 351e625d.
When the audio format is not known yet and the audio chain is still
initializing, filter reinit will fail. Normally, attempts to
reinitialize filters at this stage should be rare (e.g. user commands
editing the filter chain). But it sometimes happened with track
switching in combination with the video code calling
update_playback_speed() at arbitrary times.
Get rid of the message by not trying to change the filters for the sake
of playback speed update while decoding is still being initialized.
Until now, we've relied on the following things:
- you can send flush packets to the decoder even if it's fully flushed,
- you can send new packets to a flushed decoder,
- you can send new packers to a partially flushed decoder.
("flushing" refers to sending flush packets to the decoder until the
decoder does not return new pictures, not avcodec_flush_buffers().)
All of these are questionable. The libavcodec API probably doesn't
guarantee that these work well or at all, even though most decoders have
no issue with these. But especially with hardware decoding wrappers
(like MMAL), real problems can be expected. Isolate us from these corner
cases by handling them explicitly.
This applies to unexpected freezes or deadlocks, not e.g. normal
framedrops. The verbose messages also might remind an API user if the
API usage is incorrect, such as not calling mpv_opengl_cb_draw() when a
redraw request was issued.
This commit introduces logic to read other volumes from the same source
as the primary archive. Both .rar formats as well as 7z are supported for now.
It also changes the libarchive callback structure to be per-volume
consistent with the libarchive intenal client data array constructed
with archive_read_append_callback_data.
Added open, close and switch callbacks. Only the latter is strictly
required to make sure that the streams always start at position 0, but
leaving all volumes open can eat a lot of memory for archives with many
parts.
The nnedi3 prescaler requires a normalized range to work properly,
but the original implementation did the range normalization after
the first step of the first pass. This could lead to severe quality
degradation when debanding is not enabled for NNEDI3.
Fix this issue by passing `tex_mul` into the shader code.
Fixes#2464
vo_opengl_cb is a special case, because we somehow have to render video
asynchronously, all while "trusting" the API user to do it correctly.
This didn't quite work, and a while ago a compromise using a timeout to
prevent theoretically possible deadlocks was added.
Make it even more synchronous. Basically, go all the way, and
synchronize rendering between VO and user renderer thread to the
full extent possible.
This means the silly frame queue is dropped, and we event attempt to
synchronize the GL SwapBuffer call (via mpv_opengl_cb_report_flip()).
The changes introduced with commit dc33eb56 are effectively dropped. I
don't even remember if they mattered.
In the future, we might make all VOs fetch asynchronously from a frame
queue, which would mostly remove the differences between vo_opengl and
vo_opengl_cb, but this will take a while (if it will even be done).
Pick the correct GLSL version from the GL_SHADING_LANGUAGE_VERSION
string. Might be somewhat questionable, as we expect the minor version
number not to have leading 0s.
Should help with cases when the reported GLSL version is much higher
than the equivalent of the reported GL version. This problem was
observed in combination with GL_ARB_uniform_buffer_object, which
can't be used if the declared GLSL version is too low.
Has the same function as setting the option.
This commit changes the property in a bunch of other ways. For example
if the VO is not created, it will return the option value.
Simplifies some auto detection matters.
I _still_ don't want to remove the lazy loading mechanism, because it's
still slightly useful for filters using the hwdec APIs. My main
motivation for not always preloading them is actually that libva prints
random useless crap to the terminal with no way to prevent this.
The examples demonstrates use with optical media, which is far from
mpv's main purpose.
The authors section is a leftover from MPlayer times. There are enough
other places which reiterate how mpv is based on mplayer2/MPlayer,
copyright statements, and so on.
Hint that the linked section contains information for Windows. (Well,
that's a lie, but it has a link to the Windows section.)
Avoid implying that lines in the config file end with ';'. Also, the <>
are probably just confusing.
Deal with jittering Matroska crap timestamps. This reuses the mechanism
that is needed for frames without PTS, and adds a heuristic to it. If
the interpolated timestamp is less than 1ms away from the real one, it
might be due to Matroska timestamp rounding (or other file formats with
such rounding, or files remuxed from Matroska).
While there actually isn't much of a need to do this (audio PTS
jittering by such a low amount doesn't negatively influence much), it
helps with identifying jitter from other sources.
Instead of requiring the decoder to set the PTS directly on the
dec_audio context (including handling absence of PTS etc.), transfer the
packet PTS to the decoded audio frame. Marginally simpler, and gives
more control to the generic code.
Notes:
- Unfortunately the only way to talk to EGL from within DRM I could find
involves linking with GBM (generic buffer management for Mesa.)
Because of this, I'm pretty sure it won't work with proprietary NVidia
drivers, but then again, last time I checked NVidia didn't offer
proper screen resolution for VT.
- VT switching doesn't seem to work at all. It's worth mentioning that
using vo_drm before introduction of VT switcher had an anomaly where
user could switch to another VT and input text to it, while video
played on top of that VT. However, that isn't the case with drm_egl:
I can't switch to other VT during playback like this. This makes me
think that it's either a limitation coming from my firmware or from
EGL/KMS itself rather than a bug with my code. Nonetheless, I still
left (untestable) VT switching code in place, in case it's useful to
someone else.
- The mode_id, connector_id and device_path should be configurable for
power users and people who wish to watch videos on nonprimary screen.
Unfortunately I didn't see anything that would allow OpenGL backends
to register their own set of options. At the same time, adding them to
global namespace is pointless.
- A few dozens of lines could be shared with vo_drm (setting up VT
switching, most of code behind page flipping). I don't have any strong
opinion on this.
- Sometimes I get minor visual glitches. I'm not sure if there's a race
condition of some sort, unitialized variable (doubtful), or if it's
buggy driver. (I'm using integrated Intel HD Graphics 4400 with Mesa)
- .config and .control are very minimal.
Signed-off-by: wm4 <wm4@nowhere>
While it seemed like a pretty good idea at first, it's just a dead end
and works only in the simplest cases. While it may or may not help
slightly with audio sync mode, the display-sync mode already compensates
this in a better way. The main issue is that timestamps at this layer
are not in order, so it can look at single timestamps only.
Essentially we'd use something random, just because it's part of the srt
of traditionally used ALSA channel mappings. But each driver can do its
own things.
This doesn't let me sleep at night, so remove it.
This could accidentally change some spdif formats to AAC (because AAC is
the first on the list and will match first). spdif formats are
inherently uninterchangeable, so treat them as their own class of
formats (like int vs. float).
Might fix some issues with ao_wasapi.c.
glXCreateContextAttribsARB() by design can throw some X11 errors. We
ignore these, but we generally still print error messages to the
terminal. This was confusing/annoying users, so silence it. The stupid
part is that the Xlib error handler is global, so we have to be slightly
careful here.
They are evil and should be eradicated. Some of these were pretty dumb
anyway.
There are probably some more around in platform specific code or other
code not enabled by default on Linux.
This is based on an older patch by James Ross-Gowan. It was rebased and
cleaned up. Also, the DWM API usage present in the older patch was
removed, because DWM reports nonsense rates at least on Windows 8.1
(they are rounded to integers, just like with the old GDI API - except
the GDI API had a good excuse, as it could report only integers).
Signed-off-by: wm4 <wm4@nowhere>
This simplifies update_screen_rect a bit. Unless --fs-screen=all is
used, it will always get an HMONITOR and call GetMonitorInfo to
determine its dimensions. This will make it easier for the next few
commits to determine the colour profile and the refresh rate from the
HMONITOR.
There is a slight change in behaviour. When selecting a screen that is
out of range, such as --screen=9 on a machine with only two monitors,
the old code would silently select the last existing monitor. The new
code prints an error message and falls back to the default screen (same
as the Cocoa code.)
Signed-off-by: wm4 <wm4@nowhere>
The call to EnumDisplaySettings seems to be a relic from when MPlayer
ran on systems that didn't have GetMonitorInfo or SM_CX/CYVIRTUALSCREEN.
GetMonitorInfo was loaded dynamically, so it was possible for MPlayer to
run without it and use the values returned by EnumDisplaySettings.
These are always present in modern versions of Windows, so the values
returned from EnumDisplaySettings are always overwritten. Remove the
call to EnumDisplaySettings and assume SM_CX/CYVIRTUALSCREEN is always
present.
Signed-off-by: wm4 <wm4@nowhere>
Commit 27dc834f added it as such.
Also remove the check for glUniformBlockBinding() - it's part of an
extension, and the check glGetUniformBlockIndex() already checks whether
the extension is fully available.
Implement NNEDI3, a neural network based deinterlacer.
The shader is reimplemented in GLSL and supports both 8x4 and 8x6
sampling window now. This allows the shader to be licensed
under LGPL2.1 so that it can be used in mpv.
The current implementation supports uploading the NN weights (up to
51kb with placebo setting) in two different way, via uniform buffer
object or hard coding into shader source. UBO requires OpenGL 3.1,
which only guarantee 16kb per block. But I find that 64kb seems to be
a default setting for recent card/driver (which nnedi3 is targeting),
so I think we're fine here (with default nnedi3 setting the size of
weights is 9kb). Hard-coding into shader requires OpenGL 3.3, for the
"intBitsToFloat()" built-in function. This is necessary to precisely
represent these weights in GLSL. I tried several human readable
floating point number format (with really high precision as for
single precision float), but for some reason they are not working
nicely, bad pixels (with NaN value) could be produced with some
weights set.
We could also add support to upload these weights with texture, just
for compatibility reason (etc. upscaling a still image with a low end
graphics card). But as I tested, it's rather slow even with 1D
texture (we probably had to use 2D texture due to dimension size
limitation). Since there is always better choice to do NNEDI3
upscaling for still image (vapoursynth plugin), it's not implemented
in this commit. If this turns out to be a popular demand from the
user, it should be easy to add it later.
For those who wants to optimize the performance a bit further, the
bottleneck seems to be:
1. overhead to upload and access these weights, (in particular,
the shader code will be regenerated for each frame, it's on CPU
though).
2. "dot()" performance in the main loop.
3. "exp()" performance in the main loop, there are various fast
implementation with some bit tricks (probably with the help of the
intBitsToFloat function).
The code is tested with nvidia card and driver (355.11), on Linux.
Closes#2230
Add the Super-xBR filter for image doubling, and the prescaling framework
to support it.
The shader code was ported from MPDN extensions project, with
modification to process luma only.
This commit is largely inspired by code from #2266, with
`gl_transform_trans()` authored by @haasn taken directly.
This check disables the display-sync resample method. If the filters
convert PCM to AC3, we can still insert a filter to change speed. This
is because filters are inserted at the beginning of the filter chain.
Actually, it didn't really require that before (most work was avoided),
but some bits had to be run anyway. Separate the speed change into a
light-weight function, which merely updates already created filters, and
a heavy-weight one which messes with filter insertion.
This also happens to fix the case where the filters would "forget" the
current speed (force resampling, change speed, hit a volume control to
force af_volume insertion - it will reset speed and desync).
Since we now always run the light-weight function, remove the
af_scaletempo verbose message that is printed on speed setting. Other
than that, all setters are cheap.
Move it (in a cosmetic sense), and also move its invocation to below all
the video handling.
All other changes remain cosmetic, including moving the framedrop
calculation code, and getting rid of the video_speed_correction
variable.
For some reason, the encoder didn't like that the AVPacket already had
fields set. I'm not quite sure, but this might just be invalid API
usage. Do it as it's recommended.
We still have a sample-based buffer between filters and audio outputs.
In order to avoid cutting frames into half (which can upset receivers),
we strictly need to align the boundaries on which we cut the audio.
Update msg.c state immediately if a terminal or logging setting is set.
Until now, this was delayed until mp[v]_initialize() was called. When
using the client API, you could easily miss logged error messages, even
when logging was initialized early on by calling
mpv_request_log_messages().
(Properties can't be used for this either, because properties do not
work before mpv_initialize().)
The noframe event is logged whenever there is no new frame. This can
happen due to normal redraws, but also due to video frame queue
underflow.
The mpv_opengl_cb_report_flip() API function is currently pretty
useless, because blocking on the video frame queue is more reliable and
simpler. But at least we can log the actual vsync.
next_vsync/prev_vsync was only used to retrieve the vsync duration. We
can get this in a simpler way.
This also removes the vsync duration estimation from vo_opengl_cb.c,
which is probably worthless anyway. (And once interpolation is made
display-sync only, this won't matter at all.)
We need to effectively swap the last channel pair. See commit 4e358a96
and 5a18c5ea for details.
Doing this seems rather strange, as 7.1 just extends 5.1 with 2 new
speakers, and 5.1 doesn't need this change. Going by the HDMI standard
and the Intel HDA sources (cited in the referenced commits), it also
looks like 7.1 should simply append two channels to 5.1 as well. But
swapping them is apparently correct. This is also what XBMC does. (I
didn't find any other applications doing 7.1 PCM using the ALSA channel
map API. VLC seems to ignore the 7.1 case.) Testing reveals that at
least the end result is correct.
"Normal" ALSA 7.1 is unaffected by this, as it reports a different
(and saner) channel layout.
Instead of constructing an ALSA channel map from mpv ones from scratch,
try to find the original ALSA channel map again. Th result is that we
need to convert channel maps only in one direction. If we need to map
a mp_chmap to ALSA, we fetch the device's channel map list, convert
each entry to mp_chmap, and find the first one which fits.
This seems helpful for the following commit. For now, this only gets rid
of mapping back the trivial MONO mapping, which alone would still be
acceptable, but with other channel layout mogrifications it gets messy
fast. While we need to do something awkward to keep our channel map
reordering for VAR chmaps (which basically gives nicer output and
possibly slightly better performance), this is still the better
solution.
This reverts commit 4e358a9636.
Testing shows the channel pairs must indeed be swapped (details see
commit message of the reverted commit). Making the downmix code move
sl/sr to sdl/sdr is not an appropriate solution anymore, and it's
better to fix the unusual channel layout in ao_alsa.c directly.
(Not reverting the change in chmap.c; this is still correct.)
This affects only the display-sync code path, as for normal timing the
wakeup_pts stuff handles proper wakeup. It's probably mostly a
theoretical issue.
Discontinuities (like toggling fullscreen) can cause multiple frames to
be dropped in succession, which sounds very weird. It's better to drop
some video frames instead to compensate for larger desyncs.
We roughly base it on the maximum allowed speed changes (audio change is
"additional" to the video change to account for deviations when playing
at max. video speed change).
A hw decoder might fail to decode a frame for multiple reasons, and not
always just because decoding is impossible. We can't generally
distinguish these reasons well. Make it more tolerant by accepting
failures of 3 frames, but not more. The threshold can be adjusted by the
repurposed --vd-lavc-software-fallback option.
(This behavior was suggested much earlier in some PR, but at the time
the "proper" hwdec fallback was indistinguishable from decoding error.
With the current situation, "proper" fallback is still instantious.)
ao_alsa: attempt to fix 7.1 over HDMI
The last 2 channels of 7.1 (RLC/RRC in ALSA) were exported as sdl/sdr
instead of sl/sr (I don't even know why I chose sdl/sdr, but SL/SR
and RLC/RRC are different in the ALSA API). libsw/avresample do not
move the sl/sr channels to sdl/sdr when rematrixing, so silence was
sent for 2 channels. If my selection of sdl/sdr is essentially API
abuse, there's no reason why they should do this differently.
The mess here is really that ALSa doesn't map the HDMI layouts cleanly.
Most ALSA drivers export 7.1 in a way compatible to our expectations,
but Intel HDA/HDMI does not:
mpv/ffmpeg: fl-fr-fc-lfe-bl-br-sl-sr
ALSA/generic: FL FR FC LFE RL RR SL SR [1]
ALSA/HDMI: FL FR LFE FC RL RR RLC RRC [2]
The HDMI layout is layout 0x13 (going by CEA-861-B). The comment in
the kernel code has to be correct too. The early standard defines only
1 other layout, which replaces RLC/RRC with FRC/FLC - this probably
corresponds to what we call "7.1(wide)".
So it appears when ALSA requests RLC/RRC, we should feed it sl/sr.
To make it more complicated, Kodi/xbmc apparently also have to deal with
ALSA being special, but instead of sending sl/sr to RLC/RRC, they swap
the last two pairs of the layout, and send sl/sr to RL/RR and bl/br to
RLC/RRC. Or I might have misunderstood their code. I don't have a
7.1-capable A/V receiver, so I can't test this.
For now, go with the simpler solution, and wait until someone tests it.
If the speakers end up swapped, a completely different solution will be
needed.
[1] https://git.kernel.org/cgit/linux/kernel/git/torvalds/linux.git/tree/sound/core/pcm_lib.c?id=refs/tags/v4.3#n2434
[2] https://git.kernel.org/cgit/linux/kernel/git/torvalds/linux.git/tree/sound/pci/hda/patch_hdmi.c?id=refs/tags/v4.3#n307
These calls actually can leave the ALSA configuration space empty (how
very useful), which is why snd_pcm_hw_params() can fail. An earlier
change intended to make this non-fatal, but it didn't work for this
reason.
Backup the old parameters, so we can retry with the non-empty
configuration space. (It has to be non-empty, because the previous
setters didn't fail.)
Note that the buffer settings are not very important to us. They're
a leftover from MPlayer, which needed to write enough data to the
audio device to not underrun while decoding and displaying a video
frame. In mpv, most of these things happen asynchronously, _and_
there is a dedicated thread just for feeding the audio device, so
we should be pretty imune even against extreme buffer settings. But
I suppose it's still useful to prevent PulseAudio from making the
buffer too large, so still keep this code.
Again, this could have bad access, is unlikely, and has no bad
consequences. It's noteworthy that vlc and the ALSA PCM example both do
this first, even if they set the sample rate later.
I'm worried that not restricting the access type before restricting the
format will cause problems. While it's unlikely, it might prevent
failures in some corner cases. Also, since we by default always use
interleaved access (buggy ALSA plugins), this will have no effects at
all.
If the API doesn't list padded channel maps, but the final device
channel map is padded, and if unpadded output is not possible (unlike in
the somewhat similar dmix case), then we shouldn't apply the channel
count mismatch fallback in the beginning. Do it after channel map
negotiation instead.
Doesn't matter much; effectively this prevents just log spam in some
cases where the map is legitimately padded. Normally this is really
only needed for the dmix ALSA case. (See git blame for details.)
The uninit() function was called twice if the uninit() function failed
(once by init(), once by vd_lavc.c code), which caused crashes due to
double-free. (This failure is a corner case, and all other hwdec
backends appear to handle this case gracefully.)
I do not think this code should be able to deal with uninit() being
called other than once. Guarantee that it's called exactly once.
Quoting MSDN: "Notifies the Desktop Window Manager (DWM) to opt in to or
out of Multimedia Class Schedule Service (MMCSS) scheduling while the
calling process is alive.". Whatever this means. (An application can
change the scheduling priority of the window manager?)
Does this improve anything? I have no idea. Certainly this is a program
that does multimedia and graphics, so we seem to be a good match for
this.
Is it bad if we enable this even while playback is inactive or paused? I
have no idea either.
Is there a magic cargo cult function that will mark our renderer thread
as multimedia thing? I have no idea. (We use a function to enable MMCSS
for our audio thread in ao_wasapi.)
Enable it by default, but not unconditionally. Add an "auto" mode, which
disable DwmFlush if the compositor is (probably) inactive. Let's see how
this goes.
Since I accidentally enabled DwmFlush always by default (more or less)
in a previous commit touching this code, this is probably mostly just
cargo-culting, and it's uncertain whether it does anything.
Note that I still got bad vsync behavior when fullscreening mpv, and
making another window visible on the same screen. This happens even if
forcing DWM.
update_av_diff() works on the timestamps, while time_left is in real
time. When playing at not-1 speed, these are very different, and cause
the A/V difference to jitter. Fix this by scaling the expected A/V
desync to the correct range.
Commit acd5816a broke this. It was stopping playback occasionally.
Another case where the non-display-sync interpolation mode
(in->vsync_timed==true) is causing a lot of subtle issues and will be
removed soon.
PAL8 is the only format that is RGB, has only 1 component, is byte-
aligned. It was accidentally detected by the GBRP case as planar RGB.
(It would have been ok if it were gray; what ruins it is that it's
actually paletted, and the color values do not correspond to colors (but
palette entries).
Pseudo-pal formats are ok; in fact AV_PIX_FMT_GRAY is rightfully marked
as MP_IMGFLAG_YUV_P.
This didn't show up with cases where the frame pattern has a cycle of 1
or 2 like it is the case with 24-on-24 fps, or 24-on-60 fps. It did show
up with 25-on-60 fps. (We don't slow down 25 fps video to 24 on default
settings.)
In this case, we must not add the timing error of the next frame to the
A/V difference estimation of the current frame. Use the previous timing
error instead.
This is another bug resulting from the confusion about whether we
calculate parameters for the currently playing frame, or the one we're
about to queue.
Regression since commit 93db4233. I think the bit that was forgotten
here was to remove the vo_w32_config() return value completely. The VO
failed to init because that function always returned 0. This commit
removes these bits and fixes the VO.
Fixes#2434.
Yet another relatively useless option that tries to make OpenGL's sync
behavior somewhat sane. The results are not too encouraging. With a
value of 1, vsync jitter is gone on nVidia, but there are frame drops
(less than with glfinish). With 2, I get the usual vsync jitter _and_
frame drops.
There's still some hope that it might prevent too deep queuing with some
GPUs, I guess.
The timeout for the wait call is 1 second. The value is pretty
arbitrary; it should just not be too high to freeze the process (if
the GPU is un-nice), and not too low to trigger the timeout in normal
cases, even if the GPU load is very high. So I guess 1 second is ok
as a timeout.
The idea to use fences this way to control the queue depth was stolen
from RetroArch:
https://github.com/libretro/RetroArch/blob/df01279cf318e7ec90ace039d60515296e3de908/gfx/drivers/gl.c#L1856
Commit a1315c76 broke this slightly. Frame drops got counted multiple
times, and also vo.c was actually trying to "render" the dropped frame
over and over again (normally not a problem, since frames are always
queued "tightly" in display-sync mode, but could have caused 100% CPU
usage in some rare corner cases).
Do not repeat already dropped frames, but still treat new frames with
num_vsyncs==0 as dropped frames. Also, strictly count dropped frames in
the VO. This means we don't count "soft" dropped frames anymore (frames
that are shown, but for fewer vsyncs than intended). This will be
adjusted in the next commit.
vo_frame.num_vsyncs can be != 1 in some cases in normal sync mode too.
This is not a very exact fix, but in exchange it's robust. (These
vo_frame flags are way too tricky in combination with redrawing and
such.)
There were occasional shader compilation and rendering failures if FBOs
were unavailable. This is caused by the FBO caching code getting active,
even though FBOs are unavailable (i.e. dumb-mode).
Boken by commit 97fc4f.
Fixes#2432.
Fixes linker failure. How did this ever work? Apparently it did most of
the time, but apparently we just got the first case where it didn't.
Fixes#2433.
Bump it to 80, and 2 vsyncs. This is another measure against vsync
jitter. Admittedly this is a bit simplistic (and we should probably
estimate a stable estimated vsync phase instead), but for now this will
do.
It's not needed, because the additional data is not appended, but is the
total size of the audio buffer. The maximum size is the static audio
drop size (or twice, if the audio is duplicated).
Calculate the A/V difference directly in the display sync code, instead
of the awkward current way, which reuses the fields for audio sync.
We still set time_frame, because it makes falling back to audio sync
somewhat smoother.
When dropping or repeating frames, we essentially influence when the
frame after the next frame will be shown, not the next frame. This led
to dropping/repeating frames 2 times, because the A/V difference had a
delay of one frame. Compensate it with the expected value.
av_free_packet() got finally deprecated. Use av_packet_unref() instead,
which has almost the same semantics, has existed for a while, and is
available in all FFmpeg and Libav versions we support.
This was not very reliable.
In the normal vo_opengl case, this didn't deal well enough with vsync
jitter. Vsync timings can jitter quite extremely, up to a whole vsync
duration, in which case the "missed" frame counter keeps growing, even
though nothing is wrong. This behavior also messes up the A/V difference
calculation, but as long as it's within tolerance, it won't provoke
extra frame dropping/repeating. Real misses are harder to detect, and I
might add such detection later.
In the vo_opengl_cb case, this was additionally broken due to the
asynchronity between renderer and VO threads.
This is all kinds of stupid - update_avsync_after_frame() will multiply
this value with the speed at a later point, and we only update this
field for this function. (This should be refactored.)
Until recently, the channel layout code happened to catch this, but now
an explicit check is needed. Otherwise, it'd try to pad the missing
channels with NA in the channel map fallback code.
This is intended for the case when CoreAudio returns only unknown
channel layouts, or no channel layout matches the number of channels the
CoreAudio device forces. Assume that outputting stereo or mono to the
first channels is safe, and that it's better than outputting nothing.
It's notable that XBMC/kodi falls back to a static channel layout in
this case. For some messed up reason, the layout it uses happens to
match with the channel order in ALSA's/mpv's "7.1(alsa)" layout.
Share some code between ca_init_chmap() and ca_get_active_chmap(), which
also makes it look slightly nicer. No functional changes, other than the
additional log message.
If no channel layouts were determined (which can actually happen with
some "strange" devices), the selection code was falling back to mono,
because mono is always added as a fallback. This doesn't seem quite
right.
Allow a fallback to stereo too, if no channel layout could be retrieved
at all. So we always assume that mono and stereo work, if no other
layouts are available.
(I still don't know what the CoreAudio stereo layout is supposed to do.
It could be used to swap left and right channels. It could also be used
to pad/move the channels, but I have never seen that. And it can be set
to non-stereo channels, which breaks mpv. Whatever.)
The main reason is that ao_coreaudio_exclusive needs this for some OSX
devices. They want packed audio, and special-casing this in the
coreaudio code would be too much of a pain.
The maximum of channels we can support is 64 (because FFmpeg uses 64 bit
masks for channel layouts), but since struct mp_audio can get pretty
big (has static allocations of 2 pointers for each channel for planar
mode), it's less wasteful to stay lower for now.
av_get_default_channel_layout() fails with channel counts larger than 8.
The channel layout doesn't need to make sense, so pick an arbitrary
fallback.
libswresample also has options for setting the channel counts directly,
but better not introduce new concepts in the code. Also, libavresample
doesn't have these options.
Change it so that it will always return a bitmask with the correct
number of channels set if an unknown channel map is passed. This didn't
work for channel counts larger than 8, as there are not any standard
channel layouts defined with more than 8 channels (both in mpv and
FFmpeg). Instead, it returned 0.
This will help when raising the maximum allowed channel count in mpv.
Some code in af_lavrresample relies on it, more or less.
One change is that unknown channel maps won't result in lavc standard
channel layouts anymore, just a set of random speakers. This should be
fine, as the caller of mp_chmap_to_lavc_unchecked() should handle these
cases. For mp_chmap_reorder_to_lavc() this is not so clear anymore, but
should also be ok.
For normal channel maps, simply dropping NA channels is still the
correct and wanted behavior.
Currently, the mpv maximum channel count is 8. This commit is
preparation for raising this limit.
mNumberChannelDescriptions being 0 is pretty much an error, but if it
can happen, then the code checking the chmap below will trigger UB, as
chmap is not initialized at all.
Also, simplify the code a little: we never change the number of
channels, so this is just fine.
This code removes filters which can not take spdif inout. This was made
so that PCM filters are transparently dropped in spdif mode.
This entered an endless loop with:
--af=lavcac3enc:::2 --audio-channels=5.1
The forced number of output channels is incompatible with spdif. It's
trying to insert af_lavrresample as conversion filter to compensate for
it. Of course this doesn't work, which triggers the PCM filter removal.
Then it goes on normally - since the new state is exactly as before, it
will try the same thing again, forever.
Fix by reusing the retry counter, which is a very dumb but very
effective measure against these cases of filter negotiation failure. We
could try to be more clever (for example, if the removed filter is a
conversion filter, we can be sure this won't work, and error out
immediately). But better keep it simple and robust.
This speeds up redraws considerably (improving eg. <60 Hz material on a 60 Hz
monitor with display-sync active, or redraws while paused), but slightly
slows down the worst case (eg. video FPS = display FPS).
The IME is not useful for key-bindings. Handle the base ASCII chars
instead and don't show the IME window. For the sake of libmpv users, the
IME should only be disabled on mpv's GUI thread and not application-
wide.
No IME on the GUI thread should also mean that VK_PROCESSKEY will never
have to be handled, so the logic for that can be removed as well.
Older systems have certain EGL extension definitions missing. We
redefine them to make the build system easier, and because it's trivial.
But we forgot to define the EGL_LINUX_DMA_BUF_EXT identifier. (I hope
it's the only missing one.)
This makes the bitrate properties unavailable, instead of
returning 0 when:
1. No track is selected, or
2. Not enough packets have been read to have a bitrate estimate yet
Some mkv files can have this. The chapter times are still timestamps
(and thus not affected by the start time), but it misplaces the OSD
chapter ticks.
The equalizer code as it exists in vo_opengl works perfectly fine. The
situation in vo_opengl_cb is pretty different. The playback thread can't
communicate with the renderer thread synchronously (essentially to give
the API user more flexibility). So the equalizer communication has to be
done in an asynchronous way too.
There were two problems. First, the eq capabilities can change with the
pixel format, and the renderer initializes them on config only. This
means equalizers were disabled on the first config run, and options like
--video-output-levels or --brightness would not work. So we just
initialize the caps with a known superset. The player will not correctly
indicate when setting an eq doesn't work, but we're fine with it, as it
is a relatively cosmetic issue.
Second, it copied back the eq settings in the "wrong" moment (what
for?), which overwrote the settings in some cases.
Third, the eq was not reset correctly on vo init. This is needed to make
it behave the same as vo_opengl.
If the aspect ratio of the video resolution and the subtitle resolution
(the implied subtitle coordinate system) mismatch, the subtitles
obviously can't be overlayed over the video perfectly. Either you get
video that can't be covered by subtitles, or the subtitles could go
beyond the video. We don't want to stretch the subtitle to compensate
for the aspect ratio, because it would look terrible.
Until now, mpv used to fit the subtitle rectangle into the video
rectangle (letterboxing/pillarboxing). This looks odd with some sample
files with subtitle canvas being wider than the video. Also, mpc-hc
displays them in a better way. vlc stretches them, which looks bad.
While you probably can't win this game with all those broken files
around, pick the mpc-hc method to handle this.
There was a complaint that the naming is inaccurate. That's probably
right. Just use the official name instead, which is a bit clunky, but
surely correct.
Helps with broken vobsubs, which have an incorrect resolution header
set.
So we just extend the subtitle resolution to the video size, if the
video size is larger. This helps somewhat with readability, or makes
them visible at all. It should be a pretty safe change, because normally
no sub pictures are supposed to go outside of the area. It should make a
difference with broken files only.
The sample in question had a video resolution of 1888x1072, and a
subtitle resolution of 720x480. Note that always using video resolution
as subtitle resolution would break other files.
Coreaudio gives us a channel map with all entries set to
kAudioChannelLabel_Unknown. This is translated to a mpv channel map with
all channels set to NA, which has special meaning: it's an "unknown"
channel map, which acts as wildcard and can be converted from/to any
channel layout. Not really what we want.
I've got this with USB audio, playing stereo. The multichannel layout
consisted of 2 unknown channels, while the stereo channel map was
stereo (as expected).
Note that channel maps with _some_ NA entries are not affected by this,
and must still work.
If the device returns an unexpected number of channels instead of the
requested count on init, don't immediately error out. Instead, look if
there's a channel map with the given number of channels.
If there isn't, still error out, because we don't want to guess the
channel layout.
It's great that the new algorithm supports multiple placebo iterations
and all, but it's really not necessary and hurts performance in the
general case for the sake of the 0.1% that actually pause the screen
and look for minute differences.
Signed-off-by: wm4 <wm4@nowhere>
Reportedly fixes operation with "USB connected Parasound ZDAC v.2". (OSX
and USB audio sure is not nice at all.)
This might be perceived as hang by some users, so it's quite possible
that this will have to be adjusted again somehow.
Fixes#2409.
This reverts commit c10fb4ce9f.
This is already done in vo_wayland.c:resize,324 doing it here makes the window bigger before the video resizes showing a black area while dragging the border.
Apparently this function caused weird problems to me. I have no idea
why. The usage of the function looks perfectly fine to me, and even
rounding issues can be excluded. In any case, getting rid of this solved
my problem, and makes the code actually more readable.
The previous commit moved the av_frame_unref() after the got_picture
check. This accidentally also deferred the software fallback
reinitialization to until a software picture was decoded (instead of the
exact time of the fallback), which is not ideal.
Just rely on the fact that calling av_frame_unref() on a frame is ok
even if nothing was decoded.
Commit 12cd48a8 started setting the hwdec_failed field even if hwdec was
not active, and because it also checked this field even if hwdec was not
active, broke decoding forever.
Fix this, and also avoid a memory leak or API misuse by releasing the
decoded picture. Passing an unreleased frame to the decoder has as far
as I know no defined effects.
This fixes initial decoding of some samples. See #1341.
According to Libav devs, this should be considered a libavcodec bug, but
as it's hard to fix, here we go.
Adds support for AV_PIX_FMT_GBRP9, AV_PIX_FMT_GBRP10, AV_PIX_FMT_GBRP12,
AV_PIX_FMT_GBRP14, AV_PIX_FMT_GBRP16, AV_PIX_FMT_GBRAP, and
AV_PIX_FMT_GBRAP16.
(Not that it matters, because nobody uses these anyway.)
This VA_FOURCC isn't even defined by latest drivers, so I'm just
assuming it doesn't exist and never existed. For planar 4:2:0,
VA_FOURCC_YV12 is normally preferred, and there's even a VA_FOURCC_IYUV
for 4:2:0 with unswapped planes.
0.34 and 0.35 don't have the buffer API, such as vaAcquireBufferHandle.
This is only needed for the EGL interop, but why bother staying
compatible for such old things (0.36 was released over a year ago).
We also can drop some minor compatibility ifdeffery.
MKV files can very well start with timestamps other than 0. While mpv
has support for such files in general, and demux_lavf enables this
feature, demux_mkv didn't export a start time.
Implement this by simply reading the first cluster timestamp. This in
turn is done by reading 1 block. While we don't need the block for this
prupose at all, it's the easiest way to get the cluster timestamp read
correctly without code duplication. In theory this could be wrong, and
a packet could start at a much later time, but in practice this won't
happen.
This commit also adds an option to disable this feature. It's not
documented because nobody should use it. (But I happen to have a need
for this.)
Let's hope this doesn't confuse client API users too much. It's still
the best solution to get rid of corner cases where it actually return
the wrong timestamp on start, and then suddenly jump.
matplotlib is pathetically slow, which makes using stats-conv.py to view
dumps longer than a few seconds a huge pain.
pyqtgraph is slightly faster than matplotlib. Other than that, it seems
to be worse in every aspect (at least for plotting), but such is life.
When seeking, the current frame will have the wrong timestamp, until the
seek is done and a new frame from the seek target is shown. We still use
the "old" frame for redrawing during seeks. This frame has to be
explicitly marked with still=true in order not to confuse the
interpolation queue. If this is not done, it will ignore frames until a
frame with approximately the same timestamp as the "old" frame is
reached. (Does this mean interpolation handles timestamp resets
incorrectly? I have no idea.)
Of course we also have to clear possibly queued frames on seeks.
Also, in pausing, explicitly let the frame redraw.
This adjustment is supposed to improve the audio speed calculation in
case of unexpected desync. The flipped sign made it actually worse,
although the total impact of this bug was very minor.
Small adjustments to the playback speed use swr_set_compensation()
to stretch the audio as it is required. But since large adjustments
are now handled by actually reinitializing libswresample, the small
adjustments get rounded off completely with typical frame sizes.
Compensate for this by accounting for the rounding error and keeping
track of fractional samples that should have been output to achieve
the correct ratio.
This fixes display sync mode behavior, which requires these adjustments
to be relatively accurate.
Newer nVidia drivers support EGL, but they seem to work badly,
apparently don't support some needed features or not in a form we want
(such as swap control), and vdpau interop is not available. Disable it
by default, because I'm tired of explaining this issue.
Can be reverted as soon as nVidia release working drivers.
The libavcodec h264 decoder contains some idiotic code with unknown
purpose (no sample or explanation known that necessitates its
existence), that causes the AVCodecContext.get_format callback to be
invoked at a time when hwaccels can't be initialized. By definition, the
get_format callback is supposed to initialize hwaccels (another idiotic
thing now part of the API, but different story). This causes hwdec
initialization sometimes to fail (WolfensteinTwitch.mp4): the first
get_format callback will mark it as failed, so the second get_format
(the "proper" normal one) will not bother restoring the state, and hwdec
init fails.
While this should be fixed in libavcodec (good luck with that), it's
quite easy to workaround.
This affects the subtitle preroll mode during seeking. It could matter
somewhat with insane files with ten-thousands of subtitle events, which
now seem to pop up, and will avoid packet queue overflow.
swr/avresample_set_compensation() was made for small speed adjustments.
Non-documentation says it should be used for changes not larger than 1%,
so reinitialize the sampler if the change is larger than that.
This fixes a regression since commit f4d62da8. The original code run
vo_cocoa_config_window() once without creating the window, which had the
effect that the last part of the function was run at least once before
the actual window was created. Fix the regression by moving it to before
the window is created.
The regression itself is hard to describe. One test case: start mpv from
a fullscreened terminal window. It should switch to another desktop,
with the mpv window visible. This didn't happen anymore.
mpv_opengl_cb_uninit_gl() still needs the OpenGL context. It makes calls
to free OpenGL objects. Strictly speaking, this is probably unnecessary,
because the OpenGL context is destroyed afterwards (implicitly freeing
all related objects). But mpv_opengl_cb_uninit_gl() does not require the
destruction of the OpenGL context, and thus has to free resources
manually.
It's also true that OpenGL normally simply ignores API calls (or returns
errors) if no context is set, but doing it properly is cleaner.
That makeCurrent() can be called in the destructor is explicitly allowed
and recommended for freeing GL resources in the Qt docs.
This fixes a mpv error message on exit.
Thanks to rcombs, ffmpeg now properly supports DASH and we can
remove our hacks for it and use it by default whenever
available. If you don't like this for whatever reason, you
can get the "normal" streams back with --ytdl-format=best .
Closes#579Closes#1321Closes#2359
If video EOF happens during playback restart, and audio is syncing, and
the demuxer packet queue overflows (i.e. no new packets will be read),
then it could happen that the player accidentally enters sleeping, and
continues playing anything only after e.g. user input wakes it up.
libass 0.13.0 breaks this due to removal of fontconfig from its core
(instead, fontconfig is one possible backend, and pattern lookup is
apparently not possible anymore).
FFmpeg supports all formats the old subreader code does, and is better
at it. On the other hand, subreader.c's probing is bad and can lead to
false positives easily.
swr_set_compensation() changes the apparent sample rate on the fly (who
would have guessed). It is thus very well-suited for adjusting audio
speed on the fly during playback (like needed by the display-sync mode).
It skips the relatively slow resampler reinitialization.
If this doesn't work (libswresample soxr backend), then fall back to the
old method.
The previous commit handled not falling back to normal decoding if the
AO was reloaded (I think...), and this tries to re-engage spdif pass-
through if it was previously falling back to normal decoding (e.g.
because it temporarily switched to an audio device incapable of
passthrough).
The stop command didn't always stop. In this case, opening a HLS URL and
then sending "stop" during loading would actually make it fallback to
parsing it as a playlist, and then continued to play the playlist items.
(This corner case makes several unfortunate factors come together to
produce this really odd behavior.)
Another issue is that the "stop" was not always explicitly set. This
could be a problem when sending several commands at once. Only the
"quit" command should have priority over the "stop" command, so this is
still checked.
The Copyright file explains the whole license mess. The earlier change
was apparently confusing, because the link reading "details" merely
linked to the GPLv2 license instead of explaining anything. In fact, I
meant to link to the Copyright file in the first place.
Use the first encountered packet PTS/DTS as base, instead of the last
one. This does not add the amount of frames buffered in the codec to the
PTS offset, and thus is better.
Also, don't add the frame time if there was no decoded frame yet. The
first frame should obviously have the timestamp of the first packet
(going by this heuristic).
While b-frame reordering limits the maximum required number to around
16, the number of additionally buffered frames can be much higher.
Guess when this actually matters? (For the libavcodec MMAL wrapper.)
Useless. Sometimes it might be useful to make some extremely broken
files work, but on the other hand --no-correct-pts is sufficient for
these cases.
While we still need some of the code for AVI, the "auto" mode in
particular inflated the size of the code.
This can't be handled correctly at all. Other cases when the decoder
might drop a frame (such as completely failing to decode a frame) will
shift timestamps by a frame, and it can't be avoided.
While we could maybe find a better way to handle this with libavcodec's
main decoders, this seems to be much harder if it should work with
certain HW decoders, which don't passthrough the DTS field (such as
MMAL). Another problem are .avi files with b-frames. So just leave it
as it is.
The manpage entry explains this.
(Maybe this option could be always enabled and removed. I don't quite
remember what valid use-cases there are for just disabling audio
entirely, other than that this is also needed for audio decoder init
failure.)
Make the code a bit more uniform. Always build a "dummy" audio output
list before probing, which means that opening preferred devices and
pure auto-probing is done with the same code. We can drop the second
ao_init() call.
This also makes the next commit easier, which wants to selectively
fallback to ao_null. This could have been implemented by passing a
different requested audio output list (instead of reading it from
MPOptions), but I think it's better if this rather special feature
is handled internally in the AO code. This also makes sure the AO
code can handle its own options (such as the audio output list) in
a self-contained way.
This was used only by the timestamp sorting code, which is a fallback
for avi files (as well as avi-muxed mkv files). This was supposed to
prevent accumulating timestamps in case the decoder consumes more
packets than it outputs frames (i.e. frames are dropped). This didn't
work very well (timestamps could be off by a large amount), the
estimation of the delay was fragile, and the interdependencies with the
decoder were annoying, so kill it.
This parameter has been unused for years (the last flag was removed in
commit d658b115). Get rid of it.
This affects the general VO API, as well as the vo_opengl backend API,
so it touches a lot of files.
The VOFLAGs are still used to control OpenGL context creation, so move
them to the OpenGL backend code.
Make handling of metadata slightly more generic, and add reading of the
"PERFORMER" fields. There are some more fields, but for now let's leave
it at this.
TRACK-specific PERFORMER fields have to be read from the per-chapter
metadata (somewhat obscure).
Fixes#2328.
This gets rid of an old hack, VOFLAG_HIDDEN. Although handling of it has
been sane for a while, it used to cause much pain, and is still
unintuitive and weird even today.
The main reason for this hack is that OpenGL selects a X11 Visual for
you, and you're supposed to use this Visual when creating the X window
for the OpenGL context. Which means the X window can't be created early
in the common X11 init code, but the OpenGL code needs to do something
before that. API-wise you need separate functions for X11 init and X11
window creation. The VOFLAG_HIDDEN hack conflated window creation and
the entrypoint for resizing on video resolution change into one
function, vo_x11_config_vo_window(). This required all platform backends
to handle this flag, even if they didn't need this mechanism.
Wayland still uses this for minor reasons (alpha support?), so the
wayland backend must be changed before the flag can be entirely removed.
If interpolation is enabled, then this causes heavy artifacts if done
while unpaused. It's preferable to allow a latency of a few frames for
the change to take full effect instead. If this is done paused, the
frame is fully redrawn anyway.
This reverts commit d11184a256.
Unfortunately, there was a lot of unexpected resistance.
Do note that this is still extremely slow, crappy, etc.
Note that vo_x11.c was further edited. Compared to the removed vo_x11.c,
an additional ~200 lines of code was removed in order to simplify it. I
tried to strip it down as much as possible. In particular, support for
odd non-32 bit formats (24, 16, 15, 8 bit) is dropped.
Closes#2300.
Don't print the URL that is opened twice. stream.c and stream_lavf.c
each printed it once. Remove the logging from stream_lavf.c, and move
the log call to a more interesting point.
The vf_format suboption is replaced with --video-output-levels (a global
option and property). In particular, the parameter is removed from
mp_image_params. The mechanism is moved to the "video equalizer", which
also handles common video output customization like brightness and
contrast controls.
The new code is slightly cleaner, and the top-level option is slightly
more user-friendly than as vf_format sub-option.
This AVPacket field was a hack against the fact that the duration field
was merely an int (too small for things like subtitle durations). Newer
libavcodec drops this field and makes duration 64 bit.
This essentially reverts commit 009dfbe3. FFmpeg VideoToolbox support
is being wacky, and can cause major issues, such as not being able
to decode a single frame. (E.g. by playing a .ts file. This should be
fixed in FFmpeg eventually.)
This is not a straight revert of the commit; just a functional one. We
keep the slightly simpler code structure.
It doesn't deal with VDA at all anymore. Rename it to hwdec_osx.c. Not
using hwdec_videotoolbox.c, because that would give it the longest
source path in this project yet. (Also, this code isn't even
VideoToolox-specific, other than the name of the pixel format used.)
VideoToolbox is preferred. Now that FFmpeg released 2.8, there's no
reason to support VDA anymore. In fact, we had a bug that made VDA not
useable with older FFmpeg versions in some newer mpv releases.
VideoToolbox is supported even on slightly older OSX versions, and if
not, you still can run mpv without hw decoding.
Definitely not needed anymore, and fixes a crash in some weird corner-
cases.
The extradata freeing is apparently still needed, though. (Because a
codec context can be opened again, which makes no sense, but ok.)
This can happen with USB audio. There was already code for this, but
something in mpv and ALSA changed - and now the old code is not
necessarily triggered anymore. It probably depends on the exact
situation.
This could sometimes cause crashes in hotplug events. (Apparently in
cases when CoreAudio changes its state asynchronously, or such.)
CA_GET_STR() does not set the string if there was an error, so errors
have to be strictly checked before using it.
There are at least 2 ways of using VAAPI without X11 (Wayland, DRM).
Remove the X11 requirement from the decoder part and the EGL interop.
This will be used by a following commit, which adds Wayland support.
The worst about this is the decoder part, which includes a bad hack for
using the decoder without any VO interop (also known as "vaapi-copy"
mode). Separate the X11 parts so that they're self-contained. For the
EGL interop code we do something similar (it's kept slightly simpler,
because it essentially only has to translate between our silly
MPGetNativeDisplay abstraction and the vaGetDisplay...() call).
It looks like my hope that we can unconditionally include EGL headers in
the OpenGL code is not coming true, because OSX does not support EGL at
all. So I prefer loading the VAAPI EGL/GL specific extensions manually,
because it's less of a mess. Partially reverts commit d47dff3f.
While EGL 1.4 seemed a bit ambiguous about this to me, it actually says
quite clearly that core functions are not supported with
eglGetProcAddress() in the following paragraph.
Normally, we prefer GLX on X11. But for the VAAPI EGL interop, we
obviously want EGL. Since nvidia does not provide EGL with desktop GL
yet, we can leave it to the autoprobing. Just make sure some failure
messages don't unnecessarily show up in the nvidia case.
This breaks VAAPI GLX interop by default, but I don't care much. If
you use --hwdec=auto (which you should if you want hw decoding), this
should fallback to vaapi-copy instead.
Probe the surface format, and check whether it's really something we
support. This also does a complete check whether the EGL interop works
at all (the only way to find this out is actually running this code).
Also, support YV12. Under some circumstances, vaapi (with Intel
drivers) can be made to use this format.
Unfortunately, the Intel drivers show some very weird behavior, which
is hopefully a bug. insane_hack() provides a very evil workaround (see
comments). A proper solution might be passing the hw format as part of
mp_image_params, but as long as hw surfaces appear to be able to change
the format on the fly, attempting this is probably not worth the extra
complexity and likely fragility. The hack allows us to pretend that
there is sane behavior for now.
This makes it much faster if the surface is really mapped from GPU
memory. It's slightly slower than system memcpy if used on system
memory. We don't really know definitely in which type of memory
it's located, so we use the GPU memcpy in all cases.
Fixes#2317.
Make the GPU memcpy from the dxva2 code generally useful to other parts
of the player.
We need to check at configure time whether SSE intrinsics work at all.
(At least in this form, they won't work on clang, for example. It also
won't work on non-x86.)
Introduce a mp_image_copy_gpu(), and make the dxva2 code use it. Do some
awkward stuff to share the existing code used by mp_image_copy(). I'm
hoping that FFmpeg will sooner or later provide a function like this, so
we can remove most of this again. (There is a patch, bit it's stuck in
limbo since forever.)
All this is used by the following commit.
Broken by commit d47dff3f. If something is going to include EGL.h,
header_fixes.h has to know. This definitely affected vo_rpi, and
probably affects wayland builds (with x11egl didabled) as well.
Checking and resetting the VAImage.buf field is non-sense, even if it
happened to work out in the normal case. buf is actually freed when
vaDestroyImage() is called (not quite intuitive), and we need an extra
field to know whether vaReleaseBufferHandle() has to be called.
Printing "Using vaDeriveImage()" every frame is too verbose, so raise
the log level.
mp_image strides are in int and not unsigned int; fix this. It's not
like it actually matters, though.
Finish a comment.
Add the upstream symbolic names as comments. Normally, these should be
defined in libdrm's drm_fourcc.h header. But DRM_FORMAT_R8 and
DRM_FORMAT_GR88 are not defined anywhere, except in the kernel userland
headers of Linux 4.3 (!). We don't want mpv to depend on bleeding-edge
Linux kernel headers, so this will have to do.
Also, just for completeness, add fourccs for the 3 and 4 channel
formats. I didn't manage to test them, though.
Reduces the amount of hardcoded assumptions about the layout
drastically. (Now adding yuv420 support would be just adjusting an if,
if you ignore the other problems, such as determining the hw format at
all early enough.)
Don't call eglDestroyImageKHR() on the same ID possibly more than once.
Clear the image reference on termination, or we would leak up to 1 image
per VO recreation.
Should work much better than the old GLX interop code. Requires Mesa 11,
and explicitly selecting the X11 EGL backend with:
--vo=opengl:backend=x11egl
Should it turn out that the new interop works well, we will try to
autodetect EGL by default.
This code still uses some bad assumptions, like expecting surfaces to be
in NV12. (This is probably ok, because virtually all HW will use this
format. But we should at least check this on init or so, instead of
failing to render an image if our assumption doesn't hold up.)
This repo was a lot of help: https://github.com/gbeauchesne/ffvademo
The kodi code was also helpful (the magic FourCC it uses for
EGL_LINUX_DRM_FOURCC_EXT are nowhere documented, and
EGL_IMAGE_INTERNAL_FORMAT_EXT as used in ffvademo does
not actually exist).
(This is the 3rd VAAPI GL interop that was implemented in this player.)
The VAAPI EGL interop code will need access to the X11 Display. While
GLX could return it from the current GLX context, EGL has no such
mechanism. (At least no standard one supported by all implementations.)
So mpv makes up such a mechanism.
For internal purposes, this is very rather awkward solution, but it's
needed for libmpv anyway.
These extensions use a bunch of EGL types, so we need to include the EGL
headers in common.h to use our GL function loader with this.
In the future, we should probably require presence of the EGL headers to
reduce the hacks. This might be not so simple at least with OSX, so for
now this has to do.
Surfaces used by hardware decoding formats can be mapped exactly like a
specific software pixel format, e.g. RGBA or NV12. p->image_params is
supposed to be set to this format, but it wasn't.
(How did this ever work?)
Also, setting params->imgfmt in the hwdec interop drivers is pointless
and redundant. (Change them to asserts, because why not.)
This is a pseudo-OpenGL extension for letting libmpv query native
windowing system handles from the API user. (It uses the OpenGL
extension mechanism because I'm lazy. In theory it would be nicer to let
the user pass them with mpv_opengl_cb_init_gl(), but this would require
a more intrusive API change to extend its argument list.)
The naming of the extension and associated function was unnecessarily
Windows specific (using "D3D"), even though it would work just fine for
other platforms. So deprecate the old names and introduce new ones. The
old ones still work.
This turns the old scalers (inherited from MPlayer) into a pre-
processing step (after color conversion and before scaling). The code
for the "sharpen5" scaler is reused for this.
The main reason MPlayer implemented this as scalers was perhaps because
FBOs were too expensive, and making it a scaler allowed to implement
this in 1 pass. But unsharp masking is not really a scaler, and I would
guess the result is more like combining bilinear scaling and unsharp
masking.
Usually, libavcodec ignores errors reported by the hardware decoding
API, so it's not like we can actually escape if the hardware is somehow
acting up.
For normal fallback purposes, or if parts of the hw decoding API which
we actually check fails, we do this by setting and checking the
hwdec_failed flag anyway.
This can happen if the hw decoder allocates padded surfaces (e.g.
mod16), but the VPP output surface was allocated with the exact size.
Apparently VPP requires matching input and output sizes, or it will add
artifacts. In this case, it added mirrored pixels to the bottom few
pixels.
Note that the previous commit should have fixed this. But it didn't
work, while this commit does.
Fixes#2320.
If not set, VPP will use the whole surface. This is a problem if the
surfaces are padded, and especially if the surfaces are padded by
different amounts.
This is an attempt to fix#2320, but it appears to do nothing at all.
The comment was largely outdated, and described the old situation when
we used a "violent" fallback by making get_buffer2 fail completely.
Also, for the case when the hw decoder initialization succeeded (in
get_format), but get_buffer2 for some reason requests something
unexpected, we also can fallback more gracefully and in the same way.
2015-09-23 14:20:50 +02:00
423 changed files with 24698 additions and 20029 deletions
die "Unable to find iconv which should be part of standard compilation environment. Aborting. If you really mean to compile without iconv support use --disable-iconv."
fi
_soundcard_header=sys/soundcard.h
_check_snd=yes
check_statement_libs "sys/soundcard.h" auto SYS_SOUNDCARD_H sys/soundcard.h
test $(defretval) = yes && _soundcard_header=sys/soundcard.h && _check_snd=no
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Martin Herkt