| Commit message (Collapse) | Author | Age | Files | Lines |
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Mapping of spdif formats was imperfect. Since the first format on the
list is somehow AAC, it was returned first, which is confusing, because
CoreAudio calls all spdif formats AC3. Since the spdif formats have some
rather arbitrary, reverse mapping the formats didn"t actually work
either. Fix by explicitly ignoring these when spdif is used.
Also, don't forget to set the samplerate in ca_asbd_to_mpformat(), or it
will work only in some cases.
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This field is basically deprecated or for convenience only, and
this code doesn't need it.
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This is basically a hack for drivers which prevent the mpv DXVA2 decoder
glue from working if OpenGL is in fullscreen mode.
Since it doesn't add any "hard" new API to the client API, some of the
code would be required for a true zero-copy hw decoding pipeline, and
sine it isn't too much code after all, this is probably acceptable.
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Since we still read-back (and don't have hard plans on changing this),
this doesn't have much of an advantage.
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Preparation for the following commit.
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The individual library versionsd are pretty useless. This will actually
tell us at least the git hash or git tag of the FFmpeg build.
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Instead of only reloading the demuxer, reopen the stream as well.
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For the sake of removing the separate stream/demuxer loading code.
This could probably be reimplemented in some other way, but I have no
DVB hardware for testing. The most preferred way would be making DVB to
not quit, and just rerun the stream selection.
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The final goal is making opening the demuxer and opening the stream the
same operation.
Stream dumping is a rather uninteresting feature, but has a small
number of vocal users, and it's easy to keep.
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When seeking to a different position, and seeking takes long, the OSD
might get redrawn. This means that the VO will receive a request to
redraw an old frame using whatever the previous PTS was. This breaks the
interpolation logic: the old frame will be added to the queue, and then
the next frames (with lower PTS if you seeked backwards) are not drawn
as the logic assumes they're past frames.
Fix this by using the non-interpolation code path when redrawing after a
seek reset, and no "real" frame has been drawn yet.
It's a recent regression caused by the redrawing code simplification.
The old code simply sent a VOCTRL for redrawing the frame, and the VO
had to deal with retaining the old frame on its own.
This is a hack as in there's probably a better solution.
Fixes #2097.
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bother vo_vdpau.c, which actually uses these times.
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Use the newer internal GL backend API.
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vo_xv.c is the only place where these things are used.
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Less code, and avoids a black flash on start.
In theory it could happen that we map the window, and then don't have a
frame to draw - but mapping the window is done in the exact moment we
have a new frame to display.
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This flags stuff tried to be too clever - if there are overlapping flags
(e.g. exclusive or combined flags), the one matching with most bits has
to be chosen.
This fixes logging of the seek command. E.g. "relative" and "absolute"
overlap to make them exclusive, but "relative" was always printed as it
happened to match first.
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The value 1 is useful in some contexts, but not such a good choice
otherwise.
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This is not the most theoretically perfect solution, ideally we could
check to see if the frame in question has already been rendered
somewhere in the queue and then avoid re-rendering it, at the cost of a
few extra lines of code. But I don't think the performance trade-off is
dramatic enough here.
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draw_image_timed is renamed to draw_frame. struct frame_timing is
renamed to vo_frame. flip_page_timed is merged into draw_frame (the
additional parameters are part of struct vo_frame). draw_frame also
deprecates VOCTRL_REDRAW_FRAME, and replaces it with a method that
works for both VOs which can cache the current frame, and VOs which
need to redraw it anyway.
This is preparation to making the interpolation and (work in progress)
display sync code saner.
Lots of other refactoring, and also some simplifications.
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This should make interpolation work much better in general, although
there still might be some side effects for unusual framerates (eg. 35 Hz
or 48 Hz). Most of the common framerates are tested and working fine.
(24 Hz, 30 Hz, 60 Hz)
The new code doesn't have support for oversample yet, so it's been
removed (and will most likely be reimplemented in a cleaner way if
there's enough demand). I would recommend using something like robidoux
or mitchell instead of oversample, though - they're much
smoother for the common cases.
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For now, this is trivial (and actually redundant). The future display
sync code will make better use of it. The main point is that the new
internal API pretty much makes this transparent to the vo_opengl
interpolation code.
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Now the VO can request a number of future frames with the last parameter
of vo_set_queue_params(). This will be helpful to fix the interpolation
code.
Note that the first frame (after playback start or seeking) will usually
not have any future frames (to make seeking fast). Near the end of the
file, the number of future frames will become lower as well.
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With all the reordering etc. that can go on in this filter, it's useful
to see what upmix/downmix it's actually performing.
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Yep, the FFmpeg API can return this.
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I don't think most of these suggestions are overly helpful. Just get rid
of them.
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Until now, it only used the hash from the previous configure run,
instead of trying to get the latest hash. The "old" build system did
this correctly - we just have to use the existing logic in version.sh.
Since waf supports separate build dirs, extend version.sh with an
argument for setting the path of version.h.
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These are definitely not per-track.
(Maybe we should just provide a script or such which pretty-prints
"native" property output.
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Scrolling right should increase volume, not decrease it.
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May help with (supposedly) bad drivers, which can put the device into
some sort of broken state when trying to set a different physical
format. When the previous format is restored, it apparently recovers.
This might make the change-physical-format suboption more robust.
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We can be pretty sure that AudioUnit will remix for us.
Before this commit, we usually upmixed to stereo, because the
stereo and multichannel layouts were the only whitelisted ones.
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If the black bars appeared on the left/right borders, panscan=1 didn't
make the video cover the whole screen.
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Signed-off-by: wm4 <wm4@nowhere>
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Again. With the old OpenGL interop dropped, this probably works better
than vaapi-copy now. Last time we defaulted to vaapi-copy, because the
OpenGL interop could swap U/V planes and other stupid crap. We'll see.
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Work around that FFmpeg doesn't distinguish between surface and cropped
size. The decoder always aligns the surface size to something
"convenient" (e.g. 16 for h264), and to get to the correct cropped size,
the output image's width/height is reduced. Using the cropped size
instead of the real surface size breaks the libva API in certain cases,
so we simply store and use the original size in our per-surface struct.
(If data is cropped on the left/top borders, hw decoding will simply
display these - FFmpeg doesn't let us do better.)
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In theory, this code path avoids a copy. In practice, it never seems
to get enabled at all. But it does have potential for weird bugs or
performance issues (like being mapped from non-cacheable memory),
so kill it.
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Some window managers let you change the fullscreen state of any window
using a key combination. For example, on XFWM you can use Alt+F11 and
on Compiz you can configure a key combination with the
"Extra WM actions" plugin.
With this change mpv will handle these fullscreen state changes. So, if
you enter into fullscreen mode using the WM's shortcut and then you use
mpv's fullscreen toggle, you will get back into window mode.
Merges PR #2081.
Signed-off-by: wm4 <wm4@nowhere>
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If the EditionFlagOrdered is set, chapters without ChapterTimeEnd make
no sense. Ordered chapters will play the chapters in the order they
appear, but will play the ranges the chapters cover. So if the end time
is missing, the range is incomplete and it's not clear what should be
played. If you assume the start of the next chapter as end time, the
ordered flag will have no observable effect, so that's not a useful
assumption.
This fixes playback of a file which (apparently) had the
EditionFlagOrdered set accidentally, with normal chapters.
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Pointed out by a certain wiiaboo.
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At least Matroska files have a "forced" flag (in addition to the
"default" flag). Export this flag. Treat it almost like the default
flag, but with slightly higher priority.
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The "FrameRate" element is probably deprecated (it's greyed out in the
"spec", and described as "Informational only" in bold). Normally files
use DefaultDuration. In fact, the FrameRate field was preferred over
DefaultDuration for determining framerate if present. Do not do this and
rely on DefaultDuration only.
Also, if no framerate is set, do not assume PAL (25 FPS). Such a
fallback makes little sense and will cause more problems than it solves.
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Some of the ASCII art makes sense (like the lines starting with "|"),
but these do not make any sense to me and just look annoying.
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"mpv --ao=wasapi:help" on Linux gave "Option ao doesn't exist.".
Completely misleading and stupid.
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We want to distinguish actual errors, and just aborting the program
intentionally.
Also be a bit more careful with handling the wait() exit status: do not
called WEXITSTATUS() without checking WIFEXITED() first.
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This probably fixes the regression introduced with commit 6147bcce.
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mpv usually sets the terminal to non-canonical mode (which in particular
disables line buffering). But the old mode is restored if the process is
not foregrounded. This is supposed to make mpv behave nicer when it is
backgrounded.
getch2_poll() enables canonical mode. Unfortunately, this was only
called after the poll timeout elapsed, so non-canonical mode is first
enabled after about a second after program start. Fix this by moving the
poll call before the timeout.
(As far as we're aware, there's no event-based way to determine when the
FD's process group changes, thus we're polling.)
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And use it for the mono case. This is slightly more formal and will make
it easier to add more such cases.
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DVD/BD menu support never worked right, and are a pain to maintain. In
particular, DVD menus never actually worked correctly, because
highlights were not rendered correctly. Fixing this requires major
effort, which I'm not interested to spend.
Most importantly, the requirement to switch streams without losing the
DVD/BD state caused major weirdness in the playback core. It was
implemented by somehow syncing the playback state to the DVD/BD
implementation (in stream_dvdnav.c etc.), and then reloading the demuxer
without destroying and recreating the stream. This caused a bunch of
special-cases which I'm looking forward to remove.
For now, don't just remove everything related to menu support and just
disable it. If someone volunteers, it can be restored (i.e. rewritten)
in a reasonable way. If nobody volunteers soon, it goes.
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No need to define extra types.
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Replace all the check macros with function calls. Give them all the
same case and naming schema.
Drop af_fmt2bits(). Only af_fmt2bps() survives as af_fmt_to_bytes().
Introduce af_fmt_is_pcm(), and use it in situations that used
!AF_FORMAT_IS_SPECIAL. Nobody really knew what a "special" format
was. It simply meant "not PCM".
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Having a big switch() is simpler.
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Audio formats used a semi-clever schema to encode the properties of the
PCM encoding as bitfields into the format integer value.
The af_fmt_change_bits() implementation becomes a bit weird, but it's
an improvement to the rest of the code.
(I've always disliked it, so why not get rid of it.)
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This is actually the last line of code outside of format.c/h which still
tries to fiddle with the format bitfields.
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So whoever (nobody?) would want to deal with this broken and obscure AO
for an obscure audio API could add support for some more channel
layouts.
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The X11 video output was removed recently and is a difference
from mplayer. That's why it should be documented in the
mplayer-changes document.
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It only causes additional maintenance work.
Even if you wanted to have a fallback, it's probably better to use
--vo=sdl or so.
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This saves us the trouble of interleaving the audio data for
no reason.
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