Not specifying a --start or using --start=100% with
--play-direction=backward usually does not work. The demuxer gets no
packets and immediately enters EOF state, which then hangs because
backward playback mode neither considers this mode, nor propagates the
EOF.
As far as demuxer implementations are concerned, this behavior is OK and
even wanted. Seeking near the end with SEEK_FORWARD set is allowed not
to return any packets (so a normal relative forward seek as done by the
user would end playback). Seeking exactly to the end or past it without
SEEK_FORWARD set is probably also sane.
Another vaguely related issue is that a backward seek during playback
start does not "establish" the demux position correctly: if stream A
hits the next keyframe and seeks back, while stream B has not had a
chance to read a packet yet, then stream B will never try to read from
the old position. The effect is that stream B (and thus playback) will
effectively miss the seek target. This is "random" because it depends on
the order and number of packet read calls made by the decoders.
Fixing this is probably hard, and requires extending the already complex
state machine with more states, so turn the manpage into a TODO list for
now.
Raw audio formats can be accessed sample-wise, and logically audio
packets demuxed from it would contain only 1 sample. This is
inefficient, so raw audio demuxers typically "bundle" multiple samples
in one packet.
The problem for the demuxer cache and backward playback is that they
need properly aligned packets to make seeking "deterministic". The
requirement is that if you read some packets, and then seek back, you
eventually see the same packets again. demux_raw basically allowed to
seek into the middle of a previously returned packet, which makes it
impossible to make the transition seamless. (Unless you'd be aware of
the packet data format and cut them to make it seamless, which is too
complex for such a use case.)
Solve this by always aligning seeks to packet boundaries. This reduces
the seek accuracy to the arbitrarily chosen packet size. But you can use
hr-seek to fix this. The gain from not making raw audio an awful special
case pays in exchange for this "stupid" suggestion to use hr-seek.
It appears this also fixes that it could and did seek into the middle of
the frame (not sure if this code was ever tested - it goes back to
removing the code duplication between the former demux_rawaudio.c and
demux_rawvideo.c).
If you really cared, you could introduce a seek flag that controls
whether the seek is aligned or not. Then code which requires
"deterministic" demuxing could set it. But this isn't really useful for
us, and we'd always set the flag anyway, unless maybe the caching were
forced disabled.
libavformat's wav demuxer exhibits the same issue. We can't fix it (it
would require the unpleasant experience of contributing to FFmpeg), so
document this in otions.rst. In theory, this also affects seek range
joining, but the only bad effect should be that cached data is
discarded.
See manpage additions. This is a huge hack. You can bet there are shit
tons of bugs. It's literally forcing square pegs into round holes.
Hopefully, the manpage wall of text makes it clear enough that the whole
shit can easily crash and burn. (Although it shouldn't literally crash.
That would be a bug. It possibly _could_ start a fire by entering some
sort of endless loop, not a literal one, just something where it tries
to do work without making progress.)
(Some obvious bugs I simply ignored for this initial version, but
there's a number of potential bugs I can't even imagine. Normal playback
should remain completely unaffected, though.)
How this works is also described in the manpage. Basically, we demux in
reverse, then we decode in reverse, then we render in reverse.
The decoding part is the simplest: just reorder the decoder output. This
weirdly integrates with the timeline/ordered chapter code, which also
has special requirements on feeding the packets to the decoder in a
non-straightforward way (it doesn't conflict, although a bugmessmass
breaks correct slicing of segments, so EDL/ordered chapter playback is
broken in backward direction).
Backward demuxing is pretty involved. In theory, it could be much
easier: simply iterating the usual demuxer output backward. But this
just doesn't fit into our code, so there's a cthulhu nightmare of shit.
To be specific, each stream (audio, video) is reversed separately. At
least this means we can do backward playback within cached content (for
example, you could play backwards in a live stream; on that note, it
disables prefetching, which would lead to losing new live video, but
this could be avoided).
The fuckmess also meant that I didn't bother trying to support
subtitles. Subtitles are a problem because they're "sparse" streams.
They need to be "passively" demuxed: you don't try to read a subtitle
packet, you demux audio and video, and then look whether there was a
subtitle packet. This means to get subtitles for a time range, you need
to know that you demuxed video and audio over this range, which becomes
pretty messy when you demux audio and video backwards separately.
Backward display is the most weird (and potentially buggy) part. To
avoid that we need to touch a LOT of timing code, we negate all
timestamps. The basic idea is that due to the navigation, all
comparisons and subtractions of timestamps keep working, and you don't
need to touch every single of them to "reverse" them.
E.g.:
bool before = pts_a < pts_b;
would need to be:
bool before = forward
? pts_a < pts_b
: pts_a > pts_b;
or:
bool before = pts_a * dir < pts_b * dir;
or if you, as it's implemented now, just do this after decoding:
pts_a *= dir;
pts_b *= dir;
and then in the normal timing/renderer code:
bool before = pts_a < pts_b;
Consequently, we don't need many changes in the latter code. But some
assumptions inhererently true for forward playback may have been broken
anyway. What is mainly needed is fixing places where values are passed
between positive and negative "domains". For example, seeking and
timestamp user display always uses positive timestamps. The main mess is
that it's not obvious which domain a given variable should or does use.
Well, in my tests with a single file, it suddenly started to work when I
did this. I'm honestly surprised that it did, and that I didn't have to
change a single line in the timing code past decoder (just something
minor to make external/cached text subtitles display). I committed it
immediately while avoiding thinking about it. But there really likely
are subtle problems of all sorts.
As far as I'm aware, gstreamer also supports backward playback. When I
looked at this years ago, I couldn't find a way to actually try this,
and I didn't revisit it now. Back then I also read talk slides from the
person who implemented it, and I'm not sure if and which ideas I might
have taken from it. It's possible that the timestamp reversal is
inspired by it, but I didn't check. (I think it claimed that it could
avoid large changes by changing a sign?)
VapourSynth has some sort of reverse function, which provides a backward
view on a video. The function itself is trivial to implement, as
VapourSynth aims to provide random access to video by frame numbers (so
you just request decreasing frame numbers). From what I remember, it
wasn't exactly fluid, but it worked. It's implemented by creating an
index, and seeking to the target on demand, and a bunch of caching. mpv
could use it, but it would either require using VapourSynth as demuxer
and decoder for everything, or replacing the current file every time
something is supposed to be played backwards.
FFmpeg's libavfilter has reversal filters for audio and video. These
require buffering the entire media data of the file, and don't really
fit into mpv's architecture. It could be used by playing a libavfilter
graph that also demuxes, but that's like VapourSynth but worse.
ytdl_hook.lua essentially uses these headers to implement parts of DASH.
Hopefully the FFmpeg DASH demuxer gets usable at some point, and/or mpv
gets a proper DASH demuxer. In any case, these EDL hacks could get
removed as soon as they get unnecessary and too annoying.
Used by the next commit. It mostly exposes part of mp4_dash
functionality. It actually makes little sense other than for ytdl
special-use. See next commit.
The ytdl wrapper can resolve web links to playlists. This playlist is
passed as big memory:// blob, and will contain further quite normal web
links. When playback of one of these playlist entries starts, ytdl is
called again and will resolve the web link to a media URL again.
This didn't work if playlist entries resolved to EDL URLs. Playback was
rejected with a "potentially unsafe URL from playlist" error. This was
completely weird and unexpected: using the playlist entry directly on
the command line worked fine, and there isn't a reason why it should be
different for a playlist entry (both are resolved by the ytdl wrapper
anyway). Also, if the only EDL URL was added via audio-add or sub-add,
the URL was accessed successfully.
The reason this happened is because the playlist entries were marked as
STREAM_SAFE_ONLY, and edl:// is not marked as "safe". Playlist entries
passed via command line directly are not marked, so resolving them to
EDL worked.
Fix this by making the ytdl hook set load-unsafe-playlists while the
playlist is parsed. (After the playlist is parsed, and before the first
playlist entry is played, file-local options are reset again.) Further,
extend the load-unsafe-playlists option so that the playlist entries are
not marked while the playlist is loaded.
Since playlist entries are already verified, this should change nothing
about the actual security situation.
There are now 2 locations which check load_unsafe_playlists. The old one
is a bit redundant now. In theory, the playlist loading code might not
be the only code which sets these flags, so keeping the old code is
somewhat justified (and in any case it doesn't hurt to keep it).
In general, the security concept sucks (and always did). I can for
example not answer the question whether you can "break" this mechanism
with various combinations of archives, EDL files, playlists files,
compromised sites, and so on. You probably can, and I'm fully aware that
it's probably possible, so don't blame me.
This commit adds an extension to mpv EDL, which basically allows you to
do the same as --audio-file, --external-file, etc. in a single EDL file.
This is a relatively quick & dirty implementation. The dirty part lies
in the fact that several shortcuts are taken. For example, struct
timeline now forms a singly linked list, which is really weird, but also
means the other timeline using demuxers (cue, mkv) don't need to be
touched. Also, memory management becomes even worse (weird object
ownership rules that are just fragile WTFs). There are some other
dubious small changes, mostly related to the weird representation of
separate streams.
demux_timeline.c contains the actual implementation of the separate
stream handling. For the most part, most things that used to be on the
top level are now in struct virtual_source, of which one for each
separate stream exists. This is basically like running multiple
demux_edl.c in parallel. Some changes could strictly speaking be split
into a separate commit, such as the stream_map type change.
Mostly untested. Seems to work for the intended purpose. Potential for
regressions for other timeline uses (like ordered chapters) is probably
low. One thing which could definitely break and which I didn't test is
the pseudo-DASH fragmented EDL code, of which ytdl can trigger various
forms in obscure situations. (Uh why don't we have a test suite.)
Background:
The intention is to use this for the ytdl wrapper. A certain streaming
site from a particularly brain damaged and plain evil Silicon Valley
company usually provides streams as separate audio and video streams.
The ytdl wrapper simply does use audio-add (i.e. adding it as external
track, like with --audio-file), which works mostly fine. Unfortunately,
mpv manages caching completely separately for external files. This has
the following potential problems:
1. Seek ranges are rendered incorrectly. They always use the "main"
stream, in this case the video stream. E.g. clicking into a cached range
on the OSC could trigger a low level seek if the audio stream is
actually not cached at the target position.
2. The stream cache bloats unnecessarily. Each stream may allocate the
full configured maximum cache size, which is not what the user intends
to do. Cached ranges are not pruned the same way, which creates disjoint
cache ranges, which only use memory and won't help with fast seeking or
playback.
3. mpv will try to aggressively read from both streams. This is done
from different threads, with no regard which stream is more important.
So it might happen that one stream starves the other one, especially if
they have different bitrates.
4. Every stream will use a separate thread, which is an unnecessary
waste of system resources.
In theory, the following solutions are available (this commit works
towards D):
A. Centrally manage reading and caching of all streams. A single thread
would do all I/O, and decide from which stream it should read next. As
long as the total TCP/socket buffering is not too high, this should be
effective to avoid starvation issues. This can also manage the cached
ranges better. It would also get rid of the quite useless additional
demuxer threads. This solution is conceptually simple, but requires
refactoring the entire demuxer middle layer.
B. Attempt to coordinate the demuxer threads. This would maintain a
shared cache and readahead state to solve the mentioned problems
explicitly. While this sounds simple and like an incremental change,
it's probably hard to implement, creates more messy special cases,
solution A. seems just a better and simpler variant of this. (On the
other hand, A. requires refactoring more code.)
C. Render an intersection of the seek ranges across all streams. This
fixes only problem 1.
D. Merge all streams in a dedicated wrapper demuxer. The general demuxer
layer remains unchanged, and reading from separate streams is handled as
special case. This effectively achieves the same as A. In particular,
caching is simply handled by the usual demuxer cache layer, which sees
the wrapper demuxer as a single stream of interleaved packets. One
implementation variant of this is to reuse the EDL infrastructure, which
this commit does.
All in all, solution A would be preferable, because it's cleaner and
works for all external streams in general.
Some previous commit tried to prepare for implementing solution A. This
could still happen. But it could take years until this is finally
seriously started and finished. In any case, this commit doesn't block
or complicate such attempts, which is also why it's the way to go.
It's worth mentioning that original mplayer handles external files by
creating a wrapper demuxer. This is like a less ideal mixture of A. and
D. (The similarity with A. is that extending the mplayer approach to be
fully dynamic and without certain disadvantages caused by the wrapper
would end up with A. anyway. The similarity with D. is that due to the
wrapper, no higher level code needs to be changed.)
EDLs can be provided either as external file, or "inline" as a big
edl:// URL. There is no difference between them, except if it's loaded
from an external file, there is some weird filename sanitation going on
(see fix_filenames() in demux_edl.c). It seems this is intended to be a
security mechanism, but probably makes no sense at all.
Note that playlists are allowed to access anything locally. One
difference to playlists is that the EDL code lacks the "security"
mechanism when accessing playlist entries (see handling of the
playlist_entry.stream_flags field - EDL would need something similar),
so don't remove that, as I'm unaware of the exact consequences.
Extending the client-allocated mpv_opengl_drm_params struct
constituted a break of ABI that could cause UB.
Create a clean break by deprecating "drm_params" and related structs
and enum values, and replacing it with "drm_params_v2".
Also fix some comments and code that wrongly assumed that open could
return any other negative number than -1 for failure.
This commit updates the libmpv version to 1.104
Originally, vo_gpu/vo_opengl considered the case of Nvidia proprietary
drivers, which required vdpau/GLX, and Intel open source drivers, which
require vaapi/EGL. Since window creation and GPU context creation are
inseparable in mpv's internal API, it had to pick the correct API very
early, or hardware decoding wouldn't work. "x11probe" was introduced for
this reason. It created a GLX context (without showing the window yet),
and checked whether vdpau was available. If yes, it used GLX, if not, it
continued probing x11/EGL. (Obviously it couldn't always fail on GLX
without vdpau, which is why it was a separate "probe" backend.)
Years passed, and now the situation is different. Vdpau is dead. Nvidia
drivers and libavcodec now provide CUDA interop, which requires EGL, and
fixes some of the vdpau problems. AMD drivers now provide vaapi, which
generally works better than vdpau. Intel didn't change.
In particular, vaapi provides working HEVC Main10 support. In theory, it
should work on vdpau too, with quality reduction (no 10 bit surfaces),
but I couldn't get it to work.
So always prefer EGL. And suddenly hardware decoding works. This is
actually rather important, because HEVC is unfortunately on the rise,
despite shitty encoders and unoptimized decoders. The latter may mean
that hardware decoding works better than libavcodec.
This should have been done a long, long time ago.
The "program" property could switch between TS programs. It was rather
complex and rather obscure (even if you deal with TS captures, you
usually don't need it). If anyone actually needs it (did anyone ever
attempt to even use it?), it should be rewritten. The demuxer should
export a program list, and the frontend should handle the "cycling"
logic.
Linux analog TV support (via tv://) was excessively complex, and
whenever I attempted to use it (cameras or loopback devices), it didn't
work well, or would have required some major work to update it. It's
very much stuck in the analog past (my favorite are the frequency tables
in frequencies.c for analog TV channels which don't exist anymore).
Especially cameras and such work fine with libavdevice and better than
tv://, for example:
mpv av://v4l2:/dev/video0
(adding --profile=low-latency --untimed even makes it mostly realtime)
Adding a new input layer that targets such "modern" uses would be
acceptable, if anyone is interested in it. The old TV code is just too
focused on actual analog TV.
DVB is rather obscure, but has an active maintainer, so don't remove it.
However, the demux/stream ctrl layer must go, so remove controls for
channel switching. Most of these could be reimplemented by using the
normal method for option runtime changes.
This removes anything related to DVD/BD/CD that negatively affected the
core code. It includes trying to rewrite timestamps (since DVDs and
Blurays do not set packet stream timestamps to playback time, and can
even have resets mid-stream), export of chapters, stream languages,
export of title/track lists, and all that.
Only basic seeking is supported. It is very much possible that seeking
completely fails on some discs (on some parts of the timeline), because
timestamp rewriting was removed.
Note that I don't give a shit about optical media. If you want to watch
them, rip them. Keeping some bare support for DVD/BD is the most I'm
going to do to appease the type of lazy, obnoxious users who will care.
There are other players which are better at optical discs.
stream_dvd.c contained large amounts of ancient, unmaintained code,
which has been historically moved to libdvdnav. Basically, it's full of
low level parsing of DVD on-disc structures.
Kill it for good. Users can use the remaining dvdnav support (which
basically operates in non-menu mode). Users have reported that
libdvdread sometimes works better, but this is just libdvdnav's problem
and not ours.
This is a straightforward parallel implementation of error diffusion
algorithms in compute shader. Basically we use single work group with
maximal possible size to process the whole image. After a shift
mapping we are able to process all pixels column by column.
A large ring buffer are allocated in shared memory to speed things up.
However the size of required shared memory depends linearly on the
height of video window (or screen height in fullscreen mode). In case
there is no enough shared memory, it will fallback to `--dither=fruit`.
The maximal allowed work group size is hardcoded as 1024. Ideally we
could query `GL_MAX_COMPUTE_WORK_GROUP_INVOCATIONS`. But for whatever
reason, it seems most high end card from nvidia and amd support only
the minimal required value, so I guess we can stick to it for now.
I assume (but cannot confirm) that VA-AP-API is in fact a typo, because
most if not all search engine results related to it are from mpv's manual
page.
By changing this to VA-API and clarifying that this requires VA-API support
on a system to use it, we can hopefully make it clear to unsuspecting
Windows users that this is not the filter they're looking for.
Concerns #6690.
This allows to select the drm mode using a string specification. You
can either select the the preferred mode, the mode with the highest
resolution, by specifying WxH[@R] or by its index in the list of modes
as before.
This was implemented by using OPT_STRING_VALIDATE for drm-mode,
instead of OPT_INT. Using a string here also prepares for future
additions to drm-mode that aim to allow specifying a mode by its
resolution.
It is useful when debugging to be able to force atomic off, or as a
workaround if atomic breaks for some user. Legacy modesetting is less
likely to break by virtue of being a less complex API.
half of the materials we used were deprecated with macOS 10.14, broken
and not supported by run time changes of the macOS theme. furthermore
our styling names were completely inconsistent with the actually look
since macOS 10.14, eg ultradark got a lot brighter and couldn't be
considered ultradark anymore.
i decided to drop the old option --macos-title-bar-style and rework
the whole mechanism to allow more freedom. now materials and appearance
can be set separately. even if apple changes the look or semantics in
the future the new options can be easily adapted.
Manual changes done:
* Merged the interface-changes under the already master'd changes.
* Moved the hwdec-related option changes to video/decode/vd_lavc.c.
Rather than the linear cd/m^2 units, these (relative) logarithmic units
lend themselves much better to actually detecting scene changes,
especially since the scene averaging was changed to also work
logarithmically.
In theory our "eye adaptation" algorithm works in both ways, both
darkening bright scenes and brightening dark scenes. But I've always
just prevented the latter with a hard clamp, since I wanted to avoid
blowing up dark scenes into looking funny (and full of noise).
But allowing a tiny bit of over-exposure might be a good thing. I won't
change the default just yet (better let users test), but a moderate
value of 1.2 might be better than the current 1.0 limit. Needs testing
especially on dark scenes.
The previous approach of using an FIR with tunable hard threshold for
scene changes had several problems:
- the FIR involved annoying hard-coded buffer sizes, high VRAM usage,
and the FIR sum was prone to numerical overflow which limited the
number of frames we could average over. We also totally redesign the
scene change detection.
- the hard scene change detection was prone to both false positives and
false negatives, each with their own (annoying) issues.
Scrap this entirely and switch to a dual approach of using a simple
single-pole IIR low pass filter to smooth out noise, while using a
softer scene change curve (with tunable low and high thresholds), based
on `smoothstep`. The IIR filter is extremely simple in its
implementation and has an arbitrarily user-tunable cutoff frequency,
while the smoothstep-based scene change curve provides a good, tunable
tradeoff between adaptation speed and stability - without exhibiting
either of the traditional issues associated with the hard cutoff.
Another way to think about the new options is that the "low threshold"
provides a margin of error within which we don't care about small
fluctuations in the scene (which will therefore be smoothed out by the
IIR filter).
Instead of desaturating towards luma, we desaturate towards the
per-channel tone mapped version. This essentially proves a smooth
roll-off towards the "hollywood"-style (non-chromatic) tone mapping
algorithm, which works better for bright content, while continuing to
use the "linear" style (chromatic) tone mapping algorithm for primarily
in-gamut content.
We also split up the desaturation algorithm into strength and exponent,
which allows users to use less aggressive desaturation settings without
affecting the overall curve.
Too many broken hardware decoders. Noticed wrong decoding of a video
file encoded with x262 on RX Vega when using VAAPI (Mesa 18.3.2).
Looks fine with swdec and a cheap hardware BD player.
Reverts 017f3d0674
