Refactor the PTS handling code to make it cleaner, and to separate the
bits that use PTS sorting.
Add a heuristic to fall back to DTS if the PTS us non-monotonic. This
code is based on what FFmpeg/Libav use for ffplay/avplay and also
best_effort_timestamp (which is only in FFmpeg). Basically, this 1. just
uses the DTS if PTS is unset, and 2. ignores PTS entirely if PTS is non-
monotonic, but DTS is sorted.
The code is pretty much the same as in Libav [1]. I'm not sure if all of
it is really needed, or if it does more than what the paragraph above
mentions. But maybe it's fine to cargo-cult this.
This heuristic fixes playback of mpeg4 in ogm, which returns packets
with PTS==DTS, even though the PTS timestamps should follow codec
reordering. This is probably a libavformat demuxer bug, but good luck
trying to fix it.
The way vd_lavc.c returns the frame PTS and DTS to dec_video.c is a bit
inelegant, but maybe better than trying to mess the PTS back into the
decoder callback again.
[1] https://git.libav.org/?p=libav.git;a=blob;f=cmdutils.c;h=3f1c667075724c5cde69d840ed5ed7d992898334;hb=fa515c2088e1d082d45741bbd5c05e13b0500804#l1431
These used the suffix _resync_stream, which is a bit misleading. Nothing
gets "resynchronized", they really just reset state.
(Some audio decoders actually used to "resync" by reading packets for
resuming playback, but that's not the case anymore.)
Also move the function in dec_video.c to the top of the file.
The code stopped at kAudioChannelLabel_TopBackRight and missed mapping for
5 more channel labels. These are in a completely different order that the mpv
ones so they must be mapped manually.
This includes the case when lavc decodes audio with more than 8
channels, which our audio chain currently does not support.
the changes in ad_lavc.c are just simplifications. The code tried to
avoid overriding global parameters if it found something invalid, but
that is not needed anymore.
When using PIC on x86 (eg with hardened toolchains) the ebx register is
reserverd and cannot be used in assembly code.
For vf_eq we allow the compiler to use memory as input.
For vf_noise we temoporarily borrow the ebp register.
This fixes#361.
Signed-off-by: Natanael Copa <ncopa@alpinelinux.org>
This was broken by the recent commits. Apparently realvideo timestamps
are severely mangled, and Matroska _of course_ doesn't have the sane,
umangled timestamps, but something unusable. The existing unmangling
code in demux_mkv.c didn't output proper timestamps either. Instead,
it was something weird that triggered sorting. Without sorting (it was
disabled by default recently), you'd get decreasing PTS warnings
In order to fix this, steal some code from libavcodec. Basically copy
the contents of rv34_parser.c (with some changes), which makes
everything magically work. (Maybe it would be better to use the
libavcodec parser API, but I don't want to do that just for this. An
alternative idea would be refusing to read files that have realvideo
tracks, and delegate this to demux_lavf.c, but maybe that's too redical
too.)
I wish I hadn't notice this...
Resampling with non-ancient ALSA setups works fine, so there is no
need to keep this around. Furthermore, as of writing, the default
builtin resampler used by many ALSA setups (taken from libspeex)
actually has higher quality than the default resampling modes of
avresample and swresample.
Apparently just 5 packets is not enough for the initial audio decode
(which is needed to find the format). The old code (before the recent
refactor) appeared to use 5 packets, but there were apparently other
code paths which in the end amounted to more than 5 packets being read.
The sample that failed (see github issue #368) needed 9 packets.
Fixes#368.
These packets have to be explicitly dropped, because usually libavcodec
uses 0-sized packets to flush delayed frames, meaning just passing
through these packets would have bad consequences.
Normally, libavformat doesn't output 0-sized packets anyway. But I don't
want to take any chances, so don't delete it, and just move it out of
the way to demux.c.
Incidentally this seems wrong in the configure as well because only gl-win32
depends on it instead of also making direct3d depend on this (as I did here).
If the value for --cache-on-pause is larger than --cache-min, and the
cache runs below --cache-on-pause, but above --cache-min, the logic
would demand to pause the player and then unpause immediately again.
This doesn't make much sense, and alternating the pause state in each
playloop iteration has negative consequences. Add an explicit check to
avoid this situation.
This means the code that tries to figure out the timestamp from
demuxer and decoder output is now all in dec_video.c. We set the
final timestamp on the returned image (mp_image.pts), as well as
the d_video->pts field.
The way the player uses d_video->pts field is still a bit messy. Maybe
this could be cleaned up later.
It appears PTS sorting was useful only for avi files (and VfW-muxed
mkv). Maybe it was historically also important for decoders with broken
or non-existent PTS reordering (win32 codecs?). But now that we handle
demuxers which outputs DTS only correctly, it just seems dead weight.
Disable it by default. The --pts-association-mode option is now forced
to always use the decoder's PTS value. You can still enable the old
default (auto) or force sorting. But we will probably remove this option
entirely at some point.
Make demux_mkv export timestamps at DTS when it's in VfW mode. This is
needed to get correct timestamps with the new default mode. demux_lavf
already does that.
This was needed to determine PTS from DTS, but the previous commits
make it unnecessary.
The builtin genpts hack was used for DVD, because libavformat's genpts
essentially went amok on DVD timestamp resets. See commit 65d87091 for
details.
Having the DTS directly can be useful for restoring PTS values.
The avi file format doesn't actually store PTS values, just DTS. An
older hack explicitly exported the DTS as PTS (ignoring the [I assume]
genpts generated non-sense PTS), which is not necessary anymore due to
this change.
Now the --no-correct-pts mode is like the normal mode, just with
different timestamp calculations. The semantics should be about the
same as before this commit.
Before this commit, this mode estimated the frame time by subtracting
successive packet PTS values. This is complete non-sense for video
codecs which use reordering. The code compensated frame times for these
non-sense using the FPS value, but confused the rest of the player with
non-sense jumping around timestamps. So, all in all this mode is not
very useful.
Repurpose this mode for fixed frame rate playback. This gives almost the
same behavior as the old mode with forced framerate (--fps option). The
result is simpler and often more robust.
Instead of passing the PTS as separate field, pass it as part of the
usual data structures. Basically, this removes strange artifacts from
the API. (It's not finished, though: the final decoded PTS goes through
strange paths, and filter_video() finally overwrites the decoded
mp_image's pts field with it.)
We also stop using libavcodec's reordered_opaque fields, and use
AVPacket.pts and AVFrame.pkt_pts. This is slightly unorthodox, because
these pts fields are not "really" opaque anymore, yet we treat them as
such. But the end result should be the same, and reordered_opaque is
marked as partially deprecated (it's not clear whether it's really
deprecated).
This normally shouldn't happen. It does happen with VfW-muxed mkv files,
and would normally happen with avi files (except that we force the
correct association mode using an explicit hack for avi).
This is usually prepended by warnings like:
Decreasing video pts: 0.125000 < 0.167000
and after the switch, there should be no warnings anymore.
Background: avi likes to use DTS for timestamps instead of PTS.
Basically, there are no proper timestamps in the file, only frame
numbers. And Matroska is insane and stores the made-up DTS instead
of a proper PTS. This will be handled properly later.
Original commit was implemented differently by @bugmen0t. The problem here was
that the waf API was called directly, instead of using our own check_cc (which
defaults, among other things, to non mandatory checks).
- without Utils.* always returns empty string
- subprocess doesn't need extra quoting for sh -c
- "source" is a bash'ism, not in POSIX sh
- most shell commands embed newline at the end
Apparently Cocoa precise scrolling generates a lot of spurious events with
a delta that is equal to 0.0. Make sure that they are discarded and not added
to the input queue.
Even though this only known to happen with Cocoa, I implemented this at core
level since it makes sense in general.
Fixes: #310
waf wrongly installs resources to BundleRoot/Resources instead of
BundleRoot/Contents/Resources. Will post a patch to them later. Either way,
this must worked around until the next waf patch release.
It seems like a good idea not to generate any additional network traffic
and wait times if we don't have to.
Also print the URL it's downloading from.
Note that if we require a newer waf release, there will be a problem.
Running ./bootstrap.py won't get the newest waf version anymore in case
the old version is in the source dir. Not sure how to handle this.
waf apparently only appends a pkgconfig flag if it doesn't already exist in
the lib storage. Since our configure often checks for multiple libraries in
one call we want to keep the flags as is. This is especially important to
always keep stuff like -lm in the right place.
