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.
This is not needed anymore, because we decided that the PAR of the
decoded video matters, and not the PAR of the filtered video that
arrives at the VO.
In Python sets are unordered, so iterating them after converting to a list
always leads to different results. The code iterated on them to collect all
the flags to pass to the compiler, and since the order of the flags changed,
waf would rebuild all of the C files. Seems like in Python 2 this worked as
expected by pure chance.
This commit stores the sets as lists, and converts them to sets when the set
operations are needed.
Fixes#363
When mpv is started with some video filters set (--vf is used), and
hardware decoding is requested, and hardware decoding would be possible,
but is prevented due to video filters that accept software formats only,
the fallback didn't work properly sometimes.
This fallback works rather violently: it tries to initialize the filter
chain, and if it fails it throws away the frame decoded using the
hardware, and retries with software. The case that didn't work was when
decoding the current packet didn't immediately lead to a new frame. Then
the filter chain wouldn't be reinitialized, and the playloop would stop
playback as soon as it encounters the error flag.
Fix this by resetting the filter error flag (back to "uninitialized"),
which is a rather violent, but somewhat working solution.
The fallback in general should perhaps be cleaned up later.
In theory, we can't really do this, because we don't know when a spdif
frame ends. Spdif transports compressed audio through audio setups that
were originally designed for PCM only (which includes the audio filter
chain, the AO API, most audio output APIs, etc.), and to reach this
goal, spdif pretends to be PCM. Compressed data frames are padded with
zeros, until a certain data rate is reached, which corresponds to a
pseudo-PCM format with 2 bytes per sample and 2 channels at 48000 Hz.
Of course an actual spdif frame is significantly larger than a frame
of the PCM format it pretends to be, so cutting audio data on frame
boundaries (as according to the pseudo-PCM format) merely yields an
incomplete and broken frame, not audio that plays for the desired
duration.
However, sending an incomplete frame might still be much better than the
current behavior, which simply ignores --end/--length (but still lets
the video end at the exact end time).
Should this result in trouble with real spdif receivers, this commit
probably has to be reverted.
If the --fps option was given (MPOpts->force_fps), the demuxer FPS value
was overwritten with the forced value. This was fine, since the demuxer
value wasn't needed anymore. But with the recent changes not to write to
the demuxer stream headers, we don't want to do this anymore. So
maintain the (forced/updated) FPS value in dec_video->fps.
The removed code in loadfile.c is probably redundant, and an artifact
from past refactorings.
Note that sub.c will now always use the demuxer FPS value, instead of
the user override value. I think this is fine, because it used the
demuxer's video size values too. (And it's rare that these values are
used at all.)
Stefano Pigozzi
Martin Herkt
wm4
bugmen0t
Natanael Copa
eng