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.
wm4
Kevin Mitchell
Stefano Pigozzi
Chris Mayo
fwr
Bin Jin
Kagami Hiiragi
Eric G
Alessandro Ghedini