Helper for the ab-loop-dump-cache command, see manpage additions.
This is kind of shit. Not only is this a very "special" feature, but it
also vomits more messy code into the big and already bloated demux.c,
and the implementation is sort of duplicated with the dump-cache code.
(Except it's different.) In addition, the results sort of depend what a
video player would do with the dump-cache output, or what the user wants
(for example, a user might be more interested in the range of output
audio, instead of the video).
But hey, I don't actually need to justify it. I'm only justifying it for
fun.
That's right, and it's probably not the end of it. I'll just claim that
I have no idea how to create a proper user interface for this, so I'm
creating multiple partially-orthogonal, of which some may work better in
each of its special use cases.
Until now, there was --record-file. You get relatively good control
about what is muxed, and it can use the cache. But it sucks that it's
bound to playback. If you pause while it's set, muxing stops. If you
seek while it's set, the output will be sort-of trashed, and that's by
design.
Then --stream-record was added. This is a bit better (especially for
live streams), but you can't really control well when muxing stops or
ends. In particular, it can't use the cache (it just dumps whatever the
underlying demuxer returns).
Today, the idea is that the user should just be able to select a time
range to dump to a file, and it should not affected by the user seeking
around in the cache. In addition, the stream may still be running, so
there's some need to continue dumping, even if it's redundant to
--stream-record.
One notable thing is that it uses the async command shit. Not sure
whether this is a good idea. Maybe not, but whatever. Also, a user can
always use the "async" prefix to pretend it doesn't.
Much of this was barely tested (especially the reinterleaving crap),
let's just hope it mostly works. I'm sure you can tolerate the one or
other crash?
It makes some slight sense and helps with one of the following commits.
Also rename that other function to make it sound less similar to
find_seek_target().
Always set max_bytes_bw to 0 if seekable cache is disabled, instead at
the place of its use. This is the only use of it, so the commit should
not change any behavior.
(Alternatively, this could drop the max_bytes_bw variable, use the
option directly, and keep the old code that resets it on use of the
cache is disabled.)
Until now, the following could happen: if you set a 1GB forward cache,
and a 1GB backward cache, and you opened a 2GB file, it would prune away
the data cached at the start as playback progressed past the 50% mark.
With this commit, nothing gets pruned, because the total memory usage
will still be 2GB, which equals the total allowed memory usage of 1GB +
1GB.
There are no explicit buffers (every packet is malloc'ed and put into a
linked list), so it all comes down to buffer size computations. Both
reader and prune code use these sizes to decide whether a new packet
should be read / an old packet discarded. So just add the remaining free
"space" from the forward buffer to the available backward buffer. Still
respect if the back buffer is set to 0 (e.g. unseekable cache where it
doesn't make sense to keep old packets).
We need to make sure that the forward buffer can always append, as long
as the forward buffer doesn't exceed the set size, even if the back
buffer "borrows" free space from it. For this reason, always keep 1 byte
free, which is enough to allow it to read a new packet. Also, it's now
necessary to call pruning when adding a packet, to get back "borrowed"
space that may need to be free'd up after a packet has been added.
I refrained from doing the same for forward caching (making forward
cache use unused backward cache). This would work, but has a
disadvantage. Assume playback starts paused. Demuxing will stop once the
total allowed low total cache size is reached. When unpausing, the
forward buffer will slowly move to the back buffer. That alone will not
change the total buffer size, so demuxing remains stopped. Playback
would need to pass over data of the size of the back buffer until
demuxing resume; consider this unacceptable. Live playback would break
(or rather, would not resume in unintuitive ways), even normal streaming
may break if the server invalidates the URL due to inactivity. As an
alternative implementation, you could prune the back buffer immediately,
so the forward buffer can grow, but then the back buffer would never
grow. Also makes no sense.
As far as the user interface is concerned, the idea is that the limits
on their own aren't really meaningful, the purpose is merely to vaguely
restrict the cache memory usage. There could be just a single option to
set the total allowed memory usage, but the separate backward cache
controls the default ratio of backward/forward cache sizes. From that
perspective, it doesn't matter if the backward cache uses more of the
total buffer than assigned, if the forward buffer is complete.
The last_eof field is the last known EOF state from the underlying
demuxer. Normally, seeks reset it, because obviously if seek back into
the middle of the file, you don't want last_eof to have a "wrong" value
for a short time window (until a packet is read, which would reset the
field to its correct value).
This shouldn't happen during cache seeks, because you don't touch the
underlying demuxer state.
At first, I made this change because some other work in progress
required it. It turned out that it was unnecessary, but keep the change
anyway, since it's still correct and makes the logic cleaner.
Determining how much memory something uses is very hard, especially in
high level code (yes we call code using malloc high level). There's no
way to get an exact amount, especially since the malloc arena is shared
with the entire process anyway. So the demuxer packet cache tries to get
by with an estimate using a number of rough guesses.
It seems this wasn't quite good. In some ways, it was too optimistic, in
others it seemed to account for too much data. Try to get it closer to
what malloc and ta probably do. In particular, talloc adds some
singificant overhead (using talloc for mass-data was a mistake, and it's
even my fault). The result appears to match better with measured memory
usage. This is still extremely dependent on malloc implementation and so
on.
The effect is that you may need to adjust the demuxer cache limits to
cache as much data as it did before this commit. In any case, seems to
be better for me.
If the disk cache is used, the AVPacket is not used anymore and is
completely deallocated when the packet is written to disk. As a minor
bug, the AVPacket allocation itself was not freed (although it wasn't a
memory leak, since talloc still automatically freed it when the entire
demux_packet was freed). For very large caches, this could easily add up
to over hundred MB, so actually free the unneeded allocation.
Make most of the demuxer options runtime-changeable. This includes the
cache options and stream recording. The manpage documents some of the
possibly weird issues related to this.
In particular, the disk cache isn't shuffled around if the setting
changes at runtime.
Although this is not useful in general, it makes --stream-record work
with a certain video streaming service by a large dystopian company.
In the general case, this fails because normal muxing can, quite
obviously, not handle the segmented metadata in the packets. (There
isn't even a file format which could handle these, except possibly mp4.)
On the other hand, ytdl merely uses timeline/EDL to emulate DASH
streaming (unfortunately), which does not use the segmented stuff, and
stream recording will actually work.
It prints "Unexpected end of file (no clusters found)" when opening a
webm init fragment. The warning is correct, but unwanted in this case.
Add tons of kludges to avoid it.
(Actually it prints that twice, for audio and video each.)
Also, suppress another warning about a seek head entry that points
exactly to the end of the file. This is a MATROSKA_ID_CUES, which is
harmless, and, very strangely, doesn't point at any cues when you
concatenate the init fragment with a media fragment. No idea what that
crap is supposed to be.
Retarded webshit streaming protocols (well, DASH) chop a stream into
small fragments, and move unchanging header parts to an "init" fragment
to save some bytes (in the case at hand about 300 bytes for each
fragment that is 100KB-200KB, sure was worth it, fucking idiots).
Since mpv uses an even more retarded hack to inefficiently emulate DASH
through EDL, it opens a new demuxer for every fragment. Thus the
fragment needs to be virtually concatenated with the init fragment. (To
be fair, I'm not sure whether the alternative, reusing the demuxer and
letting it see a stream of byte-wise concatenated fragmenmts, would
actually be saner.)
demux_lavc.c contained a hack for this. Unfortunately, a certain shitty
streaming site by an evil company, that will bestow dytopia upon us soon
enough, sometimes serves webm based DASH instead of the expected mp4
DASH. And for some reason, libavformat's mkv demuxer can't handle the
init fragment or rejects it for some reason. Since I'd rather eat
mushrooms grown in Chernobyl than debugging, hacking, or (god no)
contributing to FFmpeg, and since Chernobyl is so far away, make it work
with our builtin mkv demuxer instead.
This is not hard. We just need to copy the hack in demux_lavf.c to
demux_mkv.c. Since I'm not _that_ much of a dumbfuck to actually do
this, remove the shitty gross demux_lavf.c hack, and replace it by a
slightly less bad generic implementation (stream_concat.c from the
previous commit), and use it on all demuxers. Although this requires
much more code, this frees demux_lavf.c from a hack, and doesn't require
adding a duplicated one to demux_mkv.c, so to the naive eye this seems
to be a much better outcome.
Regarding the code, for some reason stream_memory_open() is never meant
to fail, while stream_concat_open() can in extremely obscure situations,
and (currently) not in this case, but we handle failure of it anyway.
Yep.
Timeline (demux_timeline, for EDL and mkv ordered chapters) are a mess,
because it's the only nested demuxer case. Part of the mess comes from
shared struct stream pointers. This makes no sense, because the wrapper
(demux_timeline) doesn't have any business setting it.
Try to lessen it by not passing down streams. Instead, pass down NULL.
This prevents unintended interference, and tightens the ownership rules.
Now a demuxer always owns its stream.
On the other hand, demuxer->stream can now be NULL. This was never the
case before, and consequently there will be new bugs. At least they will
be spotted, because they've been bugs before.
struct stream is also used to access stream properties (such as whether
something is considered a network stream). Most of these have been
mirrored in struct demuxer (because the frontend has been forbidden to
access struct stream because of threading). But during initialization
was still used, so introduce an awkward struct parent_stream_info, which
unifies these.
Commit e0419fb181 changed demux_is_network_cached() to use
demuxer->stream->streaming instead of demuxer->is_network. To enable
timeline stuff to use the cache anyway, change it so that both flags can
contribute to it. The stream NULL-check is obviously due to changes in
this commit.
Instead of having to rely on the protocol matching, make a function that
creates a stream from a stream_info_t directly. Instead of going through
a weird indirection with STREAM_CTRL, add a direct argument for non-text
arguments to the open callback. Instead of creating a weird dummy
mpv_global, just pass an existing one from all callers. (The latter one
is just an artifact from the past, where mpv_global wasn't available
everywhere.)
Actually I just wanted a function that creates a stream without any of
that bullshit. This goal was slightly missed, since you still need this
heavy "constructor" just to setup a shitty struct with some shitty
callbacks.
Raise it from 8KB to 512KB.
Do this because ytdl_hook.lua generated a 40KB EDL file (from 80KB
youtube-dl JSON output), and putting it into a .m3u file for easier
debugging failed due to the size limit.
The previous commit broke audio playback (maybe this is what 4. was
about?). But it wasn't the fault of the commit; it just exposed
pre-existing issues.
If the packet queue search can't get all packets, it checked
queue->is_bof to see whether there could be previous packets. But
obviously, is_bof can be set, even if the search start packet wasn't the
first one.
This was especially observable with audio, because audio by default
needs preroll packets, and plays artifacts if they're missing.
Fix by using the BOF playback condition for this purpose too.
In theory, backward demuxing does way too much work by doing a full
cache seek every time you need to step back through a packet queue. In
theory, it would be exceedingly more efficient to just iterate backwards
through the queue, but this is not possible because I'm too stingy to
add 8 bytes per packet for backlinks. (In theory, you could probably
come up with some sort of deque, that'd allow efficient iteration into
any direction, but other requirements make this tricky, and I'm
currently too dumb/lazy to do this. For example, the queue can grow to
millions of elements, all while packet pointers need to stay valid.)
Another possibility is to "locally" seek the queue. This still has less
overhead than a full seek.
Also, it just so happens that, as a side effect, this avoids performing
range merging, which commit f4b0e7e942 "broke". That wasn't a bug at
all, but since range joining is relatively slow, avoiding it is good.
This is really just a coincidental side effect, I'm not even quite sure
why it happens this way.
There are 4 ugly things about this change:
1. To get a keyframe "before" a certain one, we recompute the target
PTS, and then subtract 0.001 as arbitrary number to "fudge" it. This
isn't the first place where this is done, and hey, it wasn't my damn
idea that MPlayer should use floats for timestamps. (At first, it even
used 32 bit timestamps.)
2. This is the first time reader_head is reset to an earlier position
outside of the seek code. This might cause conceptual problems since
this code is now "duplicated".
3. In theory, find_backward_restart_pos() needs to be run again after
the backstep. Normally, the seek code calls it explicitly. We could call
it right in the new code, but then the damn function would be recursive.
We could shuffle the code around to make it a loop, but even then
there'd be an offchance into running into an unexpected corner case (aka
subtle bug), where it would loop forever. To avoid refactoring the code
and having to think too hard about it, make it deferred - add some new
state and the check_backward_seek() function for this. Great, even more
subtle mutable state for this backwards shit.
4. I forgot this one, but I can assure you, it's bad.
Without doubt someone will have to clean up this slightly in the future
(or rip it out), but it won't be me.
This code tries to determine the "current" position, which is used as
base for the seek target when it needs to seek back more (the point is
to prevent seeking back too far). But compute_keyframe_times() almost
computes the same thing, so use that. Unfortunately needs a forward
declaration.
("Almost", because it differs in some details that should not really
matter.)
Subtitle packets with a timestamp before the seek target may overlap
with the seek target anyway. This is why this subtitle preroll crap
exists: it needs to return packets before the seek target to ensure that
the subtitle is displayed at the seek target.
This didn't always work. Maybe it's a regression, but it must have been
an old one. The breakage is triggered by heuristic that is to prevent
excessive queuing of packets in garbage files (this heuristic apparently
became immediately necessary when this preroll mechanism was
implemented).
If a video keyframe packet was found, but no audio packet yet, then
subtitle_preroll was set to 0, and since a_skip_to_keyframe was still 0,
the subtitle packet was discarded. The dumb thing is that subtitle and
video seeking is finished at this point, so the preroll crap should not
be applied at all.
Fix this by moving the preoll overflow code into the block that handles
preroll.
demux_packet.next should not be used outside of demux.c, and in this
case it's a packet that was just passed to demux.c from the outside.
demux_packet.stream is already set by the demuxer, and this is assured
by the add_packet_locked() caller.
We don't care much about this case, because backward playback can fail
terribly without a good way to detect it, so this was fine.
However, this froze in certain situations. Reading from a subtitle file
for which backward demuxing failed could make it get stuck in
demux_read_packet_async() in unthreaded mode. (That we don't support
backwards subtitle decoding anyway doesn't matter for this.)
So aggressively disable backward demuxing to prevent worse in these
situations. The behavior will still be awful, because the frontend is
still in backwards playback mode, but at least it won't freeze.
Somewhat similar to the old --cache-file, except for the demuxer cache.
Instead of keeping packet data in memory, it's written to disk and read
back when needed.
The idea is to reduce main memory usage, while allowing fast seeking in
large cached network streams (especially live streams). Keeping the
packet metadata on disk would be rather hard (would use mmap or so, or
rewrite the entire demux.c packet queue handling), and since it's
relatively small, just keep it in memory.
Also for simplicity, the disk cache is append-only. If you're watching
really long livestreams, and need pruning, you're probably out of luck.
This still could be improved by trying to free unused blocks with
fallocate(), but since we're writing multiple streams in an interleaved
manner, this is slightly hard.
Some rather gross ugliness in packet.h: we want to store the file
position of the cached data somewhere, but on 32 bit architectures, we
don't have any usable 64 bit members for this, just the buf/len fields,
which add up to 64 bit - so the shitty union aliases this memory.
Error paths untested. Side data (the complicated part of trying to
serialize ffmpeg packets) untested.
Stream recording had to be adjusted. Some minor details change due to
this, but probably nothing important.
The change in attempt_range_joining() is because packets in cache
have no valid len field. It was a useful check (heuristically
finding broken cases), but not a necessary one.
Various other approaches were tried. It would be interesting to list
them and to mention the pros and cons, but I don't feel like it.
Backwards demuxing usually seeks back back by a "random" amount (set by
a user option) when it needs new preceding packets. It turns out a past
change made these backwards seek amounts add up when it didn't need to
(i.e. subtracting the amount from the seek pos without properly
resetting it), which could possibly slow down playback as it went on.
The reason for this was that back_seek_pos was set for every stream on
every seek. This made the reset not affect other streams (in particular
streams which weren't used and never were reset, or which didn't reset
that often). But as the commit adding it showed, this is needed only to
set the initial position. So do that.
Fixes: "demux: fix initial backward demuxing state in some cases"
When packet appending sets the start of the range, it adjusts the range
by seek_preroll. Do this when packets are pruned from the start of the
range too.
(Yeah, seek_preroll handling is probably broken in some other cases. It
was halfhearted to begin with.)
The main thing this commit does is removing demux_packet.kf_seek_pts. It
gets rid of 8 bytes per packet. Which doesn't matter, but whatever.
This field was involved with much of seek range updating and pruning,
because it tracked the canonical seek PTS (i.e. start PTS) of a packet
range. We have to deal with timestamp reordering, and assume the start
PTS is the lowest PTS across all packets (not necessarily just the first
packet). So knowing this PTS requires looping over all packets of a
range (no, the demuxer isn't going to tell us, that would be too sane).
Having this as packet field was perfectly fine. I'm just removing it
because I started hating extra packet fields recently.
Before this commit, this value was cached in the kf_seek_pts field (and
computed "incrementally" when adding packets). This commit computes the
value on demand (compute_keyframe_times()) by iterating over the placket
list. There is some similarity with the state before 10d0963d85,
where I introduced the kf_seek_pts field - maybe I'm just moving in
circles. The commit message claims something about quadratic complexity,
but if the code before that had this problem, this new commit doesn't
reintroduce it, at least. (See below.)
The pruning logic is simplified (I think?) - there is no "incremental"
cached pruning decision anymore (next_prune_target is removed), and
instead it simply prunes until the next keyframe like it's supposed to.
I think this incremental stuff was only there because of very old code
that got refactored away before. I don't even know what I was thinking
there, it just seems complex. Now the seek range is updated when a
keyframe packet is removed.
Instead of using the kf_seek_pts field, queue->seek_start is used to
determine the stream with the lowest timestamp, which should be pruned
first. This is different, but should work well. Doing the same as the
previous code would require compute_keyframe_times(), which would
introduce quadratic complexity.
On the other hand, it's fine to call compute_keyframe_times() when the
seek range is recomputed on pruning, because this is called only once
per removed keyframe packet. Effectively, this will iterate over the
packet list twice instead of once, and with some locality. The same
happens when packets are appended - it loops over the recently added
packets once again. (And not more often, which would go above linear
complexity.)
This introduces some "cleverness" with avoiding calling
update_seek_ranges() even when keyframe packets added/removed, which is
not really tightly coupled to the new code, and could have been in a
separate commit.
Removing next_prune_target achieves the same as commit b275232141,
which is hereby reverted (stale is_bof flags prevent seeking before the
current range, even if the beginning of the file was pruned). The seek
range is now strictly computed after at least one packet was removed,
and stale state should not be possible anymore.
Range joining may over-allocate the index a little. It tried hard to
avoid this before by explicitly freeing the old index before creating a
new one. Now it iterates over the old index while adding the entries to
the new one, which is simpler, but may allocate twice the memory in the
worst case. It's not going to matter for anything, though.
Seeking will be slightly slower. It needs to compute the seek PTS values
across all packets in the vicinity of the seek target. The previous code
also iterated over these packets, but now it iterates one packet range
more.
Another minor detail is that the special seeking code for SEEK_FORWARD
goes away. The seeking code will now iterate over the very last packet
range too, even if it's incomplete (i.e. packets are still being
appended to it). It's fine that it touches the incomplete range, because
the seek_end fields prevent that anything particularly incorrect can
happen. On the other hand, SEEK_FORWARD can now consider this as seek
target, which the deleted code had to do explicitly, as kf_seek_pts was
unset for incomplete packet ranges.
Obviously doesn't sense with this order. The git history shows that this
comment was touched multiple times, without ever fixing it. It was
originally added in 2016, where the "for" was missing. Later, the "for"
was added, but to the wrong position.
What the fuck?
The old implementation didn't work for the OGG case. Discard the old
shit code (instead of fixing it), and write new shit code. The old code
was already over a year old, so it's about time to rewrite it for no
reason anyway.
While it's true that the old code appears to be broken, the main reason
to rewrite this is to make it simpler. While the amount of code seems to
be about the same, both the concept and the actual tag handling are
simpler. The result is probably a bit more correct.
The packet struct shrinks by 8 byte. That fact that it wasted 8 bytes
per packet for a rather obscure use case was the reason I started this
at all (and when I found that OGG updates didn't work). While these 8
bytes aren't going to hurt, the packet struct was getting too bloated.
If you buffer a lot of data, these extra fields will add up. Still quite
some effort for 8 bytes. Fortunately, it's not like there are any
managers that need to be convinced whether it's worth doing. The freedom
to waste time on dumb shit.
The old implementation attached the current metadata to each packet.
When the decoder read the packet, the packet's metadata was made
current. The new implementation stores metadata as separate list, and
requires that the player frontend tells it the current playback time,
which will be used to find the currently valid metadata. In both cases,
the objective was to correctly update metadata even if a lot of data is
buffered ahead (and to update them correctly when seeking within the
demuxer cache).
The new implementation is actually slightly more correct, because it
uses the playback time for the metadata lookup. Consider if you have an
audio filter which buffers 15 seconds (unfortunately such a filter
exists), then the old code would update the current title 15 seconds too
early, while the new one does it correctly.
The new code also simplifies mixing the 3 metadata sources (global, per
stream, ICY). We assume these aren't mixed in a meaningful way. The old
code tried to be a bit more "exact". I didn't bother to look how the old
code did this, but the new code simply always "merges" with the previous
metadata, so if a newer tag removes a field, it's going to stick around
anyway.
I tried to keep it simple. Other approaches include making metadata a
special sh_stream with metadata packets. This would have been
conceptually clean, but the implementation would probably have been
unnatural (and doesn't match well with libavformat's API anyway). It
would have been nice to make the metadata updates chapter points (makes
a lot of sense for the intended use case, web radio current song
information), but I don't think it would have been a good idea to make
chapters suddenly so dynamic. (Still an idea to keep in mind; the new
code actually makes it easier to work towards this.)
You could mention how subtitles are timed metadata, and actually are
implemented as sparse packet streams in some formats. mp4 implements
chapters as special subtitle stream, AFAIK. (Ironically, this is very
not-ideal for files. It would be useful for streaming like web radio,
but mp4 is extremely bad for streaming by design for other reasons.)
bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla
bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla
bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla
bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla
bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla
bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla
bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla
bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla
bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla
bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla
bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla
bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla
bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla
bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla bla
Some OGG web radio streams use timestamp resets when a new song starts
(you can find those Xiph's directory - other streams there don't show
this behavior). Basically, the OGG stream behaves like concatenated OGG
files, and "of course" the timestamps will start at 0 again when the
song changes. This is very inconvenient, and breaks the seekable demuxer
cache. In fact, any kind of seeking will break
This is more time wasted in Xiph's bullshit. No, having timestamp resets
by design is not reasonable, and fuck you. I much prefer the awful
ICY/mp3 streaming mess, even if that's lower quality and awful. Maybe it
wouldn't be so bad if libavformat could tell us WHERE THE FUCK THE RESET
HAPPENS. But it doesn't, and the randomly changing timestamps is the
only thing we get from its API.
At this point, demux_lavf.c is like 90% hacks. But well, if libavformat
applies this strange mixture of being clever for us vs. giving us
unfiltered garbage (while pretending it abstracts everything, and hiding
_useful_ implementation/low level details), not much we can do.
This timestamp linearizing would, in general, probably be better done
after the decoder, because then we wouldn't need to deal with timestamp
resets. But the main purpose of this change is to fix seeking within the
demuxer cache, so we have to do it on the lowest level.
This can probably be applied to other containers and video streams too.
But that is untested. Some further caveats are explained in the manpage.
Probably doesn't change anything, other than looking slightly better. In
theory, the common function has some stuff that makes it more likely
that timestamps round-trip through conversions properly, but I didn't
confirm that.
Remove the duplicated creation of the first range. Explicitly destroy
ranges, including the last one on final deinit.
It looks like this also fixes a leak of removed range structs, which was
never noticed because they're so small, and were freed on final deinit
due to having the demuxer as talloc parent.
This improves upon the previous commit too (that change should have
been part of it I guess). Sub-demuxers (demux_timeline only) now
automatically don't use the cache (like it was intended by the previous
commit). The cache is "initialized" (or disabled) last in the recursive
call chain, which is messy, but this sub demuxer stuff FUCKING SUCKS, as
mentioned in the previous commit message. This would be no problem if
the caching layer and actual demuxer implementations were separate.
Most of this change has no purpose. Might make (de-)initialization of
further cache exerpiments simpler.
It seems the so called demuxer cache wasn't really disabled for
sub-demuxers (timeline stuff). This was relatively harmless, since the
actual packet data was shared anyway via refcounting. But with the
addition of a mmap cache backend, this may change a lot.
So strictly disable any caching for sub-demuxers. This assumes that
users of sub-demuxers (only demux_timeline.c by now?) strictly use
demux_read_any_packet(), since demux_read_packet_async() will require
some minor read-ahead if a low level packet read returned a packet for a
different stream.
This requires some awkward messing with this fucking heap of trash. The
thing that is really wrong here is that the demuxer API mixes different
concepts, and sub-demuxers get the same API as decoders, and use the
cache code.
The demuxer cache tries to track the number of bytes allocated for the
cache. In addition to the packet queue, the seek index is another data
structure that roughly depends on the amount of packets cached. So the
index size should somehow be part of the total number of bytes tracking.
Until now, this was handled with KF_SEEK_ENTRY_WORST_CASE, basically a
shitty heuristic. It was a guess (and probably rather an upper bound
than a lower bound). The implementation details made it annoying, and it
was conceptually inaccurate too.
Change this, and instead simply add the index size to the total cache
size. This essentially makes it part of the backbuffer. It's nice that
this cleanly decouples it from the packet size tracking itself.
Since it's part of the backbuffer number of bytes now, packet pruning
can't necessarily free enough space in the backbuffer anymore. Before
this commit, the backbuffer consisted of packets only, so it was
possible to reduce its size to 0 by pruning all packets until the
decoder reader position, at which point a packet was accounted as
forward buffered. Now the index is added to this, and it can't be
pruned. Replace the assert() because of this changed invariant.
Someone who rams a knife into his own hand just to see what happens is
normally put in a psychiatric ward. But in software, this is acceptable
behavior. Programs are not supposed to crash just because a user did
something unreasonably dumb.
Switching tracks during backward playback is such a thing. It triggered
an assertion because the newly enabled stream was not properly
initialized for backward playback. Fix this, and make it actually work
(mostly; it still takes a "while" until playback recovers fully).
This actually makes some aspects of initialization slightly cleaner.
Not sure if this is bug-free. You _always_ make bugs when writing a
binary search from scratch (and such is the curse of C, though if I did
this in C++ it would probably end in blood). It seems to work though,
checking against the normal linear search.
It's slightly faster. Not much.
I wonder if the termination condition can be written in a nicer/elegant
way. I guess the fact that it's not a == predicate makes this slightly
messier?
The purpose of the seek index is to avoid having to iterate over the
full linked list of cached packets, which should speed up seeking. Until
now, there was an excuse of a seek index, that didn't really work.
The idea of the old index was nice: have a fixed number of entries (no
need to worry about exceeding memory requirements), which are
"stretched" out as the cache gets bigger. The size of it was 16 entries,
which in theory should speed up seeking by the factor 16, given evenly
spaced out entries. To achieve this even spacing, it attempted to "thin
out" the index by half once the index was full (see e.g. index_distance
field). In my observations this didn't really work, and the distribution
of the index entries was very uneven. Effectively, this did nothing. It
probably worked once and I can't be assed to debug my own shit code.
Writing new shit code is more fun.
Write new shit code for fun. This time it's a complete index. It's kept
in a ringbuffer (for easier LIFO style appending/removing), which is
resized with realloc if it becomes too small.
Actually, the index is not completely completely; it's still "thinned
out" by a hardcoded time value (INDEX_STEP_SIZE). This should help with
things like audio or crazy subtitle tracks (do they still create
those?), where we can just iterate over a small part of the linked
packet list to get the exact seek position. For example, for AAC audio
tracks with typical samplerates/framesizes we'd iterate about 50 packets
in the linked list.
The results are good. Seeking in large caches is much faster now,
apparently at least 1 or 2 orders of magnitude. Part of this is because
we don't need to touch every damn packet in a huge linked list (bad
cache behavior - the index is a linear memory region instead), but
"thinning" out the search space also helps. Both aspects can be easily
tested (setting INDEX_STEP_SIZE to 0, and replacing e->pts with
e->pkt->kf_seek_pts in find_seek_target()).
This does use more memory of course. In theory, we could tolerate memory
allocation failures (the index is optional and only for performance),
but I didn't bother and inserted an apologetic comment instead, have fun
with the shit code). the memory usage doesn't seem to be that bad,
though. Due to INDEX_STEP_SIZE it's bounded by the file duration too.
Try to account for the additional memory usage with an approximation
(see KF_SEEK_ENTRY_WORST_CASE). It's still a bit different, because the
index needs a single, potentially large allocation.
The search was slightly more complicated and slow than it had to be. It
didn't assume that the packet list was sorted, which is responsible for
much of this. (I think the search code was borrowed from demux_mkv.c,
which does not sort index entries.)
There was a half-hearted attempt to make it exit early, but it was
mostly ineffective.
Simplify the code based on the assumption that the list is sorted. This
will exit the search loop once the worst case candidate entry was
checked.
Mostly about the packet queue and the subtitle handling of it.
(This mess sure sounds like a good argument to give up the separate
stream queues, and using a single packet queue per cached range.)
Until now, this usually passed a single audio frame to the decoder, and
then did a backstep operation (cache seek + frame search) again. This is
probably not very efficient, especially considering it has to search the
packet queue from the "start" every time again.
Also, with most audio codecs, an additional "preroll" frame was passed
first. In these cases, the preroll frame would make up 50% of audio
decoding time. Also not very efficient.
Attempt to fix this by returning multiple frames at once. This reduces
the number of backstep operations and the ratio the preoll frames. In
theory, this should help efficiency. I didn't test it though, why would
I do this? It's just a pain. Set it to unscientific 10 frames.
(Actually, these are 10 keyframes, so it's much more for codecs like
TrueHD. But I don't care about TrueHD.)
This commit changes some other implementation details. Since we can
return more than 1 non-preroll keyframe to the decoder, some new state
is needed to remember how much. The resume packet search is adjusted to
find N ("total") keyframe packets in general, not just preroll frames.
I'm removing the special case for 1 preroll packet; audio used this, but
doesn't anymore, and it's premature optimization anyway.
Expose the new mechanism with 2 new options. They're almost completely
pointless, since nobody will try them, and if they do, they won't
understand what these options truly do. And if they actually do, they
most likely would be capable of editing the source code, and we could
just hardcode the parameters. Just so you know that I know that the
added options are pointless.
The following two things are truly unrelated to this commit, and more
like general refactoring, but fortunately nobody can stop me.
Don't set back_seek_pos in dequeue_packet() anymore. This was sort of
pointless, since it was set in find_backward_restart_pos() anyway (using
some of the same packets). The latter function tries to restrict this to
the first keyframe range though, which is an optimization that in theory
might break with broken files (duh), but in these cases a lot of other
things would be broken anyway.
Don't set back_restart_* in dequeue_packet(). I think this is an
artifact of the old restart code (cf. ad9e473c55). It can be done
directly in find_backward_restart_pos() now. Although this adds another
shitty packet search loop, I prefer this, because clearer what's
actually happening.
The size of all forward buffered packets is used to control maximum
buffering.
Until now, this size was incrementally adjusted, but had to be
recomputed on seeks within the cache. Doing this was actually pretty
expensive. It iterates over a linked list of separate memory allocations
(which are probably spread all over the heap due to the allocation
behavior), and the demux_packet_estimate_total_size() call touches a lot
of further memory locations. I guess this affects the cache rather
negatively. In an unscientific test, the recompute_buffers() function
(which contained this loop) was responsible for roughly half of the time
seeking took.
Replace this with a way that computes the buffered size between 2
packets in constant times. The demux_packet.cum_pos field contains the
summed sizes of all previous packets, so subtracting cum_pos between two
packets yields the size of all packets in between. We can do this
because we never remove packets from the middle of the queue. We only
add packets to the end, or remove packets at the beginning.
The tail_cum_pos field is needed because we don't store the end position
of a packet, so the last packet's position would be unknown. We could
recompute the "estimated" packet size, or store the estimated size in
the packet struct, but I just didn't like this.
This also removes the cached fw_bytes fields. It's slightly nicer to
just recompute them when needed. Maintaining them incrementally was
annoying. total_size stays though, since recomputing it isn't that cheap
(would need to loop over all ranges every time).
I'm always using uint64_t for sizes. This is certainly needed (a stream
could easily burn through more than 4GB of data, even if much less of
that is cached). The actual cached amount should always fit into size_t,
so it's casted to size_t for printfs (yes, I hate the way you specify
stdint.h types in printfs, the less I have to use that crap, the
better).
In ancient times, the number of packets was used to limit excessive
read-ahead. This was completely replaced by tracking the size in bytes.
The number of packets was used in debugging output only.
In one case (packet got demuxed and is added to a queue), only log
whether there were packets on this stream before. (Unknown whether it's
useful.)
In another case (queue overflow), actually count the number of packets.
It's vaguely useful, and the message with the number of packets is shown
only once after a seek reset, so it doesn't matter whether it's slow.
Some files don't start with keyframe packets. Normally, this is not
sane, but the sample file which triggered this was a cut TV capture
transport stream. And this shouldn't happen anyway.
Introduce a further heuristic: if the last seek target was before the
start of the cached data, and the start of the cache is marked as BOF
(beginning of file), then we won't find anything better. This is
possibly a bit shaky, because both seek_start and back_seek_pos weren't
made for this purpose. But I can't come up with situations where this
would actually break. (Leave this to shitty broken files I hit later.)
I also considered finding the first packet in the cache that is marked
as keyframe, i.e. the first actual seek target, and comparing it to
"first", but I didn't like it much. Well whatever.
It's a bit silly that this caused a hard freeze (and similar issues
still will). The problem is that the demuxer holds the lock and has no
reason to release it. And in general, there's a single lock for the
entire demuxer cache. Finer grained locking would probably not make much
sense. In theory status of available data and maybe certain commands to
the demuxer could be moved to separate locks, but it would raise
complexity, and you'd probably still need to get the central lock in
some cases, which would deadlock you anyway.
It would still be nice if some minor corner case in the wonderfully
terrible and complex backward demuxer state machine couldn't lock up the
player. As a hack, unlock and then immediately lock again. Depending on
the OS mutex implementation, this may give other waiters a chance to
grab the lock. This is not a guarantee (some OSes may for example not
wake up other waiters until the next time slice or something), but works
well on Linux.
