Lots of noise to remove the vfilter/vo fields from dec_video.
From now on, video filtering and output will still be done together,
summarized under struct vo_chain.
There is the question where exactly the vf_chain should go in such a
decoupled architecture. The end goal is being able to place a "complex"
filter between video decoders and output (which will culminate in
natural integration of A->V filters for natural integration of
libavfilter audio visualizations). The vf_chain is still useful for
"final" processing, such as format conversions and deinterlacing. Also,
there's only 1 VO and 1 --vf option. So having 1 vf_chain for a VO seems
ideal, since otherwise there would be no natural way to handle all these
existing options and mechanisms.
There is still some work required to truly decouple decoding.
Instead of handling this on filter chain reinit, do it directly after
the decoder. This makes the code less entangled. In particular, this
gets rid of the really weird "override params" concept in the video
filter code.
The last_format/fixed_formats have some redundance with decoder_output,
but unfortunately the latter has a slightly different use.
This is mainly a refactor. I'm hoping it will make some things easier
in the future due to cleanly separating codec metadata and stream
metadata.
Also, declare that the "codec" field can not be NULL anymore. demux.c
will set it to "" if it's NULL when added. This gets rid of a corner
case everything had to handle, but which rarely happened.
MPlayer traditionally always used the display aspect ratio, e.g. 16:9,
while FFmpeg uses the sample (aka pixel) aspect ratio.
Both have a bunch of advantages and disadvantages. Actually, it seems
using sample aspect ratio is generally nicer. The main reason for the
change is making mpv closer to how FFmpeg works in order to make life
easier. It's also nice that everything uses integer fractions instead
of floats now (except --video-aspect option/property).
Note that there is at least 1 user-visible change: vf_dsize now does
not set the display size, only the display aspect ratio. This is
because the image_params d_w/d_h fields did not just set the display
aspect, but also the size (except in encoding mode).
Approximately reverts commit 3ccac74d. This failed with some avi files,
which do pseudo-VFR by sending packets with empty frames (or repeat
frames, depending on point of view). Specifically, these packets are not
0 bytes, so they don't get skipped by libavformat, as with the usual VFR
avi hack. Instead, the packet contains a VOP with vop_coded=0, so
libavcodec will just return no frame. We could probably distinguish such
skipped frames and delayed frames by explicitly measuring the codec
delay by counting how long it takes to get the very first frame (and
then treat skips as explicit drops), but we may as well simply reinstate
the old code.
To appease to at least one semi-broken case, do not enable this logic on
the RPI, as the FFmpeg MMAL wrapper has arbitrary buffering (and MMAL
itself is asynchronous).
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).
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.
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.
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.
When showing cover art, the decoding logic pretends that the source has
an infinite number of frames. This slightly simplifies dealing with
filter data flow. It was done by feeding the same packet repeatedly to
the decoder (each decode run produces new output).
Change this by decoding once at the video initialization. This is easier
to follow, and increases robustness in case of broken images. Usually,
we try to tolerate decoding errors, so decoding normally continues, but
in this case it would just burn the CPU for no reason.
Fixes#2056.
Remove the old implementation for these properties. It was never very
good, often returned very innaccurate values or just 0, and was static
even if the source was variable bitrate. Replace it with the
implementation of "packet-video-bitrate". Mark the "packet-..."
properties as deprecated. (The effective difference is different
formatting, and returning the raw value in bits instead of kilobits.)
Also extend the documentation a little.
It appears at least some decoders (sipr?) need the
AVCodecContext.bit_rate field set, so this one is still passed through.
Remove the colorspace-related top-level options, add them to vf_format.
They are rather obscure and not needed often, so it's better to get them
out of the way. In particular, this gets rid of the semi-complicated
logic in command.c (most of which was needed for OSD display and the
direct feedback from the VO). It removes the duplicated color-related
name mappings.
This removes the ability to write the colormatrix and related
properties. Since filters can be changed at runtime, there's no loss of
functionality, except that you can't cycle automatically through the
color constants anymore (but who needs to do this).
This also changes the type of the mp_csp_names and related variables, so
they can directly be used with OPT_CHOICE. This probably ended up a bit
awkward, for the sake of not adding a new option type which would have
used the previous format.
Trying to handle such video is almost worthless, but it was requested by
at least 2 users.
If there are no timestamps, enable byte seeking by setting
ts_resets_possible. Use the video FPS (wherever it comes from) and the
audio samplerate for timing. The latter was already done by making the
first packet emit DTS=0; remove this again and do it "properly" in a
higher level.
For some reason there were two points in the code where it warned
against non-monotonic video PTS. The one in video.c triggered on PTS
going backwards or making large jumps forwards, while dec_video.c
triggered on PTS going backwards or PTS not changing. Merge them into a
single check, which warns against all cases.
This played e.g. a 1264x722 file as 1264x720. There was some code which
dropped the aspect ratio if the video (in original resolution) wasn't
scaled by more than 4 pixels. Commit 5f3c3f8c introduced this (although
I'm not really sure what the code replaced by it did).
Just remove this "feature".
DVD and Bluray (and to some extent cdda) require awful hacks all over
the codebase to make them work. The main reason is that they act like
container, but are entirely implemented on the stream layer. The raw
mpeg data resulting from these streams must be "extended" with the
container-like metadata transported via STREAM_CTRLs. The result were
hacks all over demux.c and some higher-level parts.
Add a "disc" pseudo-demuxer, and move all these hacks and special-cases
to it.
The i_bps members of the sh_audio and dev_video structs are mostly used
for displaying the average audio and video bitrates. Keeping them in
bits-per-second avoids truncating them to bytes-per-second and changing
them back lateron.
Sometimes, Matroska files store monotonic PTS for h264 tracks with
b-frames, which means the decoder actually returns non-monotonic PTS.
Handle this with an evil trick: if DTS is missing, set it to the PTS.
Then the existing logic, which deals with falling back to DTS if PTS is
broken. Actually, this trick is not so evil at all, because usually, PTS
has no errors, and DTS is either always set, or always unset. So this
_should_ provoke no regressions (famous last words).
libavformat actually does something similar: it derives DTS from PTS in
ways unknown to me. The result is very broken, but it causes the DTS
fallback to become active, and thus happens to work.
Also, prevent the heuristic from being active if PTS is merely monotonic
instead of strictly-monotonic. Non-unique PTS is broken, but we can't
fallback to DTS anyway in these cases.
The specific mkv file that is fixed with this commit had the following
fields set:
Muxing application: libebml v1.3.0 + libmatroska v1.4.1
Writing application: mkvmerge v6.7.0 ('Back to the Ground') [...]
But I know that this should also fix playback of mencoder produced mkv
files.
Also remove MSGL_SMODE and friends.
Note: The indent in options.rst was added to work around a bug in
ReportLab that causes the PDF manual build to fail.
Change how the video decoding loop works. The structure should now be a
bit easier to follow. The interactions on format changes are (probably)
simpler. This also aligns the decoding loop with future planned changes,
such as moving various things to separate threads.
This collects statistics and other things. The option dumps raw data
into a file. A script to visualize this data is included too.
Litter some of the player code with calls that generate these
statistics.
In general, this will be helpful to debug timing dependent issues, such
as A/V sync problems. Normally, one could argue that this is the task of
a real profiler, but then we'd have a hard time to include extra
information like audio/video PTS differences. We could also just
hardcode all statistics collection and processing in the player code,
but then we'd end up with something like mplayer's status line, which
was cluttered and required a centralized approach (i.e. getting the data
to the status line; so it was all in mplayer.c). Some players can
visualize such statistics on OSD, but that sounds even more complicated.
So the approach added with this commit sounds sensible.
The stats-conv.py script is rather primitive at the moment and its
output is semi-ugly. It uses matplotlib, so it could probably be
extended to do a lot, so it's not a dead-end.
Since m_option.h and options.h are extremely often included, a lot of
files have to be changed.
Moving path.c/h to options/ is a bit questionable, but since this is
mainly about access to config files (which are also handled in
options/), it's probably ok.
The tmsg stuff was for the internal gettext() based translation system,
which nobody ever attempted to use and thus was removed. mp_gtext() and
set_osd_tmsg() were also for this.
mp_dbg was once enabled in debug mode only, but since we have log level
for enabling debug messages, it seems utterly useless.
This code tried to pass a still monotonic (even if not strictly
monotonic) PTS to the player, but as a result it remained stuck on
the PTS before a reset (since the PTS was lower).
Until now, the player didn't care to drain frames on video reconfig.
Instead, the VO was reconfigured (i.e. resized) before the queued frames
finished displaying. This can for example be observed by passing
multiple images with different size as mf:// filename. Then the window
would resize one frame before image with the new size is displayed. With
--vo=vdpau, the effect is worse, because this VO queues more than 1
frame internally.
Fix this by explicitly draining buffered frames before video reconfig.
Raise the display time of the last frame. Otherwise, the last frame
would be shown for a very short time only. This usually doesn't matter,
but helps when playing image files. This is a byproduct of frame
draining, because normally, video timing is based on the frames queued
to the VO, and we can't do that with frames of different size or format.
So we pretend that the frame before the change is the last frame in
order to time it. This code is incorrect though: it tries to use the
framerate, which often doesn't make sense. But it's good enough to test
this code with mf://.
This should help fixing some issues (like not draining video frames
correctly on reinit), as well as decoupling the decoder, filter chain,
and VO code.
I also wanted to make the hardware video decoding fallback work properly
if software-only video filters are inserted. This currently has the
issue that the fallback is too violent, and throws away a bunch of
demuxer packets needed to restart software decoding properly. But
keeping "backup" packets turned out as too hacky, so I'm not doing this,
at least not yet.
This adds vf_chain, which unlike vf_instance refers to the filter chain
as a whole. This makes the filter API less awkward, and will allow
handling format negotiation better.
The --flip option flipped the image upside-down, by trying to use VO
support, or if not available, by inserting a video filter. I'm not sure
why it existed. Maybe it was important in ancient times when VfW based
decoders output an image this way (but even then, flipping an image is a
free operation by negating the stride).
One nice thing about this is that it provided a possible path for
implementing video orientation, which is a feature we should probably
support eventually. The important part is that it would be for free for
VOs that support it, and would work even with hardware decoding.
But for now get rid of it. It's useless, trivial, stands in the way, and
supporting video orientation would require solving other problems first.
So, FFmpeg/Libav requires us to figure out video timestamps ourselves
(see last 10 commits or so), but the methods it provides for this aren't
even sufficient. In particular, everything that uses AVI-style DTS (avi,
vfw-muxed mkv, possibly mpeg4-in-ogm) with a codec that has an internal
frame delay is broken. In this case, libavcodec will shift the packet-
to-image correspondence by the codec delay, meaning that with a delay=1,
the first AVFrame.pkt_dts is not 0, but that of the second packet. All
timestamps will appear shifted. The start time (e.g. the time displayed
when doing "mpv file.avi --pause") will not be exactly 0.
(According to Libav developers, this is how it's supposed to work; just
that the first DTS values are normally negative with formats that use
DTS "properly". Who cares if it doesn't work at all with very common
video formats? There's no indication that they'll fix this soon,
either. An elegant workaround is missing too.)
Add a hack to re-enable the old PTS code for AVI and vfw-muxed MKV.
Since these timestamps are not reorderd, we wouldn't need to sort them,
but it's less code this way (and possibly more robust, should a demuxer
unexpectedly output PTS).
The original intention of all the timestamp changes recently was
actually to get rid of demuxer-specific hacks and the old timestamp
sorting code, but it looks like this didn't work out. Yet another case
where trying to replace native MPlayer functionality with FFmpeg/Libav
led to disadvantages and bugs. (Note that the old PTS sorting code
doesn't and can't handle frame dropping correctly, though.)
Bug reports:
https://trac.ffmpeg.org/ticket/3178https://bugzilla.libav.org/show_bug.cgi?id=600
Using --start with files that use DTS only, or which simply have broken
PTS timestamps, would incorrectly drop frames and possibly not execute
the seek correctly.
Add yet another heuristic to detect this. The intent is that --start and
hr-seeks in general should work correctly, but in order to keep things
fast, we still want to allow frame dropping during hr-seek if there are
no problems doing so. Do this by disabling frame dropping by default,
but re-enabling it if there are no problems found for a while. As a
consequence, --start might be somewhat slower, but normal user
interaction should remain as fast as before.
Note that there's something subtle about the added code: the
has_broken_packet_pts field is checked even before the first packet is
fed to dec_video.c, so the field must not be set to 0 right on start.
It's not initially set to 0 anyway, because the heuristic requires
decoding some images before enabling frame drop anyway.
Note 2: it's not clear whether frame dropping during hr-seek really
helps; I didn't benchmark it.
wm4
Kevin Mitchell
Marcin Kurczewski
Niklas Haas
Marcoen Hirschberg
Martin Herkt