| Commit message (Collapse) | Author | Age |
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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.
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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.
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Fixes #2386.
Reminder that you should not use this crap-VO.
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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).
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Makes the spdif automagic work better on audio hotplugging.
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demux_open() kept trying to fallback to other demuxers when opening was
cancelled. This was not really a problem, but it was stupid.
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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.
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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.
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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).
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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.)
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It's obviously not needed, and only an artifact of the old PTS
determination code.
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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.
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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.
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Closes #2370
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Set format to invalid after each failed test. This way the final check
for valid format will actually fail if no formats work.
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Fixes #2382.
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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.)
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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.
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Fixes #2378.
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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.
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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.
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Why not.
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Get it out of the way in the common code. MPGLContext.dwm_flush_opt can
be removed as well as soon as the option system gets overhauled.
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This was preventing compilation on systems without X11 headers.
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This was previously trying to use the video_output_vaapi symbol despite
vo_vaapi.c being guarded by the vaapi-x11 option.
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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.
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A minor, but apparently common feature request. Fixes #2360.
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You needed to select a GL backend with the backend suboption. This was
confusing.
Fixes #2361.
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We have to disable it, or shader compilation will fail.
Fixes #2362.
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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.
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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.
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This reverts commit d11184a256ed709a03fa94a4e3940eed1b76d76f.
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.
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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.
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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.
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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.
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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.
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Get rid of the VDA specifics like naming or ancient pixel formats.
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