in->byte_level_seeks field is written and modified inside
update_bytes_read at the same time when demux_get_reader_state
is executing, which locks the demux thread mutex. This results
in a data race, reported by Thread Sanitizer when playing mp3 file
of sufficient long length.
Though, only when the output format is matroska, to avoid muxing errors.
This is quite useful when the input has ASS subtitles, as they tend to
rely on embedded fonts.
dump_cache() calls qsort() to order an array of pointers, while the
comparator forgets it's receiving pointers to pointers.
Since cache-dumping over multiple cache ranges is fairly rare, this
seems to have gone unnoticed.
Add new header which shows up as tags/metadata (associated with
--display-tags). The way this is added means it doesn't always work,
because root->meta (see code) can be NULL for some absurd reason. But it
works for the one case I intended to use it (ytdl_hook, see next
commit), though only in default configurations.
Before this commit, the user could specify a printf format string
which wasn't verified, and could result in:
- Undefined behavior due to missing or non-matching arguments.
- Buffer overflow due to untested result length.
The offending code was added at commit 103a9609 (2002, mplayer svn):
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@4566 b3059339-0415-0410-9bf9-f77b7e298cf2
It moved around but was not modified meaningfully until now.
Now we reject all conversion specifiers at the format except %%
and a simple subset of the valid specifiers. Also, we now use
snprintf to avoid buffer overflow.
The format string is provided by the user as part of mf:// URI.
Report and initial patch by Stefan Schiller.
Patch reviewed by @jeeb, @sfan5, Stefan Schiller.
This reverts commit 41243e7c4f.
This fixes image format detection. FFmpeg has an utter called "image2",
which is designed to read patterns in filenames (so you can play
something like "%*.jpg" for all jpg files in the current directory).
"image2" is not what we want; it's just broken with custom I/O like
mpv uses it, and we don't want to "accidentally" interpret filenames
as pattern. That's why mpv blacklists it.
Unfortunately, "image2" is sometimes the format that FFmpeg's probe API
returns as best match. Thus demux_lavf fails to detect the file type,
and after some more futile attempts, we end up at demux_mf, which uses
detection by file extension. (Not sure why. I guess MPlayer did that,
and foudn that sufficient.) If the file extension is wrong (which
happens a lot because apparently the world is full of idiots who don't
manage to get the most simple things right), the image "loads", but
decoding obviously fails.
There's no easy way around this. The FFmpeg API has no mechanism to
exclude a specific format from probing (like image2, which breaks stuff
for us). Out of the 5 probe functions the API provides, none can probe
a specific format or include or exclude specific formats. The main
problem is that AVInputFormat.read_probe is a private symbol.
FFmpeg itself has no problem opening such files. It turns out that it
works, because even though image2 by itself uses detection by file
extension, it uses private API to further probe the exact format. It
explicitly excludes itself to prevent recursion.
But fortunately, that also means that it's impossible to get the image2
format if no filename is passed to the prober. (No filename, no file
extension.) Apparently we pass it in because it helps in corner cases.
Until almost 3 years ago, we passed the filename only when normal
probing already failed. Restore this by this revert. It makes
incorrectly named files work. The revert also makes the (apparently
forgotten) comment above the touched line of code true again.
Yes, quite possible that this breaks some mp3s again. You can't win
with FFmpeg. Thanks FFmpeg for making us fail at opening simple image
files and/or the most widely used file format for audio.
Well, whatever. Only results in an error message being printed, because
there is no other error reporting mechanism, and the general policy is
to keep trying with the rest of the data (i.e. not report EOF).
Such files violate the specification. Unfortunately, I could not test
whether it really works correctly, since I don't have a sample at hand
that is not broken in this regard.
The header probing hacks were previously all broken. They only worked
the first time the archive file was open. Since subsequent opens (on
seek) occured in the middle of the source stream rather than at the
beginning, the stream_read_peek calls meant to retrieve the headers were
instead returning random bytes in the middle of the file.
Perhaps the worst manifestation of this was when seeking within a
multi-volume .rar archive with the "legacy" file naming pattern. If the
seek required a reopen, the fact that the archive was multi-volume would
be forgotten and the file would appear truncated terminating playback.
To solve this, only perform the header probling the first time the
archive is opened. Save the results and reuse them on subsequent
reopens. Put this in a wrapper so this is transparent to
demux_libarchive.
Instead of just picking the last tag that was encountered. The order of
the tags still depends on the file order.
This is probably wrong, and we should respect TargetTypeValue. But
despite staring at the spec, I have no idea what the hell this should
do, so fuck that.
Fixes: #7604
Unfortunately, attached pictures (from tags etc.) are treated as video
tracks. That meant --sub-create-cc-track added a CC track for them as
well. Stop doing that.
See: #7608
Replace use of .min==1 with a proper flag. This is a good idea, because
it has nothing to do with numeric limits (also see commit 9d32d62b61
for how this can go wrong).
With this, m_option.min/max are strictly used for numeric limits.
Add an infrastructure for collecting performance-related data, use it in
some places. Add rendering of them to stats.lua.
There were two main goals: minimal impact on the normal code and normal
playback. So all these stats_* function calls either happen only during
initialization, or return immediately if no stats collection is going
on. That's why it does this lazily adding of stats entries etc. (a first
iteration made each stats entry an API thing, instead of just a single
stats_ctx, but I thought that was getting too intrusive in the "normal"
code, even if everything gets worse inside of stats.c).
You could get most of this information from various profilers (including
the extremely primitive --dump-stats thing in mpv), but this makes it
easier to see the most important information at once (at least in
theory), partially because we know best about the context of various
things.
Not very happy with this. It's all pretty primitive and dumb. At this
point I just wanted to get over with it, without necessarily having to
revisit it later, but with having my stupid statistics.
Somehow the code feels terrible. There are a lot of meh decisions in
there that could be better or worse (but mostly could be better), and it
just sucks but it's also trivial and uninteresting and does the job. I
guess I hate programming. It's so tedious and the result is always shit.
Anyway, enjoy.
As an unfortunate disaster, min/max values use the type double, which
causes tons of issues with int64_t types. Anyway, OPT_BYTE_SIZE is often
used as maximum for size_t quantities, which can have a size different
from (u)int64_t.
OPT_BYTE_SIZE still uses in64_t, because in theory, you could use it for
file sizes. (demux.c would for example be capable of caching more than
2GB on 32 bit platforms if a file cache is used. Though for some reason
the accounting code still uses size_t, so that use case is broken. But
still insist that it _could_ be used this way.)
There were various inconsistent attempts to set m_option.max to a value
such that the size_t/int64_t upper limit is not exceeded. Due to the
double max field, this didn't really work correctly. Try to fix this
with the M_MAX_MEM_BYTES constant. It's a good approximation, because on
32 bit it should allow 2GB (untested, also would probably exhaust
address space in practice but whatever), and something "high enough" in
64 bit.
For some reason, clang 11 still warns. But I think this might be a clang
bug, or I'm crazy. The result is correct anyway.
Change all OPT_* macros such that they don't define the entire m_option
initializer, and instead expand only to a part of it, which sets certain
fields. This requires changing almost every option declaration, because
they all use these macros. A declaration now always starts with
{"name", ...
followed by designated initializers only (possibly wrapped in macros).
The OPT_* macros now initialize the .offset and .type fields only,
sometimes also .priv and others.
I think this change makes the option macros less tricky. The old code
had to stuff everything into macro arguments (and attempted to allow
setting arbitrary fields by letting the user pass designated
initializers in the vararg parts). Some of this was made messy due to
C99 and C11 not allowing 0-sized varargs with ',' removal. It's also
possible that this change is pointless, other than cosmetic preferences.
Not too happy about some things. For example, the OPT_CHOICE()
indentation I applied looks a bit ugly.
Much of this change was done with regex search&replace, but some places
required manual editing. In particular, code in "obscure" areas (which I
didn't include in compilation) might be broken now.
In wayland_common.c the author of some option declarations confused the
flags parameter with the default value (though the default value was
also properly set below). I fixed this with this change.
Before this commit, option declarations used M_OPT_MIN/M_OPT_MAX (and
some other identifiers based on these) to signal whether an option had
min/max values. Remove these flags, and make it use a range implicitly
on the condition if min<max is true.
This requires care in all cases when only M_OPT_MIN or M_OPT_MAX were
set (instead of both). Generally, the commit replaces all these
instances with using DBL_MAX/DBL_MIN for the "unset" part of the range.
This also happens to fix some cases where you could pass over-large
values to integer options, which were silently truncated, but now cause
an error.
This commit has some higher potential for regressions.
Change to it 1000 hours, which is "infinite" enough. (Hesitant to use
INFINITY, as that is not in the option's range. The option parser
rejects it because it causes only problems in API users and so on.)
The demuxer cache employs a strange method to make track switching
instant with caching enabled. Normally this would mean you have to wait
until the cache has played out (and you get new packets, including
packets from the newly selected track), or you have to perform a slow
high level seek (decoding video again etc.). The strange method is that
it performs a demuxer-level seek without a high level seek so it looks
like a continuous stream to the decoder, and the newly select stream
gets packets at the current playback position. This is called a refresh
seek.
This works only if some weird heuristics work. It needs a packet "unique
ID", for which it uses either dts or pts. The value must be strictly
monotonic increasing. If this doesn't work, the referesh seek can't be
executed, and the user has to wait until the end of the cache. Sometimes
there are files that simply do not work.
In the present case, there's actually a hack that we can extend. Packets
with unset position are likely generated by the parser, and the hack
which this commit touches simply attempts to make up a new (hopefully
unique) position value, even if the value itself makes no sense. It only
ha to be deterministic.
It turns out libavcodec sometimes output packets with repeating position
values. This commit tries to handle this case too with the same hack.
Fixes: #7498
Preparation for a future commit. The demuxer queues might be read from
other threads than the one to issue the seek, and passing SEEK_BLOCK
with such a seek will provide a convenient way to synchronize this.
It seems sporadic errors are possible, such as connection timeouts.
Before the recent demuxer change, the demuxer thread retried many times
even on EOF, so an error was only interpreted as EOF once the decoder
queues ran out.
Change it to use EOF only. Since this may actually lead to the demuxer
thread being "stuck" and retrying forever (depending on libavformat API
behavior), I'm also adding a heuristic to prevent this, using a random
retry counter. This should not be necessary, but libavformat cannot be
trusted. (This retrying forever could be stopped by the user, but
obviously it would still heat the CPU for a longer time if the user is
not present.)
In this case the video track has seek_start == seek_end, and due to the
"seek_start >= seek_end" condition, this was considered broken, and no
seek range was created, breaking cached seeking.
Fix this by allowing the case if they're equal, and a valid timestamp.
(NB: seeking backward in this will still jump to position 0, because it
is the video timestamp. This is unfortunately how it's supposed to work.
HR-seeks will also do this, but decode and skip the entire audio until
the seek target, so it will mostly appear to work.)
Exposed by commit b56e2efd5f. demux_timeline reported a bogus EOF if
"parallel" streams were used. If a virtual source reported EOF, it was
propagated as global EOF, without serving packets of other virtual
sources that have not ended yet.
Fix this by not reporting global EOF just because a source has not
returned a packet. Instead make the reader retry by returning no packet
and no EOF state, which will call d_read_packet() again with a different
source. Rely on the eof_reached flags to signal global EOF.
Since eof_reached is now more important, set it in a certain other case
when it apparently should have been set. do_read_next_packet()'s return
value is now ignored, so get rid of it.
This is useful with live streams, and it's not much worse than the h264
first packet hack, which reads some data anyway.
For some reason, the option wasn't even documented, so do that.
In addition, print the start time even if it's negative. That should not
be possible, but for some reason, the field is an int64_t copied from an
uint64_t so... whatever. Keeping the logging slightly more straight
forward is better anyway.
Remove some redundant fields that controlled or indicated whether the
demuxer was/could/should prefetch. Redefine how the eof/reading fields
work.
The in->eof field is now always valid, instead of weirdly being reset to
false in random situations. The in->reading field now corresponds to
whether the demuxer thread is working at all, and is reset if it stops
doing anything.
Also, I always found it stupid that dequeue_packet() forced the demuxer
thread to retry reading if it was EOF. This makes little sense, but was
probably added for files that are being appended to (running downloads).
It makes no sense, because if the cache really tried to read until file
EOF, it would encounter partial packets and throw errors, so all is lost
anyway. Plus stream_file now handles this better. So stop this behavior,
but add a temporary option that enables the old behavior.
I think checking for ds->eager when enabling prefetching never really
made sense (could be debated, but no, not really). On the other hand,
the change above exposed a missing wakeup in the backward demuxing code.
Some chances of regressions that could make it stuck in certain states
or so, or incorrect demuxer cache state reporting to the player
frontend.
A negative subtitle delay means that subtitles from the future should be
shown earlier. With muxed subtitles, subtitle packets are demuxed along
with audio and video packets. But since they are demuxed "lazily",
nothing guarantees that subtitle packets from the future are available
in time.
Typically, the user-observed effect is that subtitles do not appear at
all (or too late) with large negative --sub-delay values, but that using
--cache might fix this.
Make this behave better. Automatically extend read-ahead to as much as
needed by the subtitles. It seems it's the easiest to pass the subtitle
render timestamp to the demuxer in order to guarantee that everything is
read. This timestamp based approach might be fragile, so disable it if
no negative sub-delay is used.
As far as the player frontend part is concerned, this makes use of the
code path for external subtitles, which are not lazily demuxed, and may
already trigger waiting.
Fixes: #7484
Subtitle tracks are usually "lazy" (ds->eager=false), There are a number
of weird special cases associated with it. One of them is that they have
some sort of "temporary" EOF (to signal that there isn't a packet right
now, and the decoder should not block playback by waiting for more
packets). In a the next commit, I want to call mark_stream_eof() in case
of (some) of these temporary EOFs.
The problem is that mark_stream_eof() also calls the functions touched
by this commit. Basically they shouldn't do any complex work due to
these temporary EOFs (because they might happen very often). It turns
out that lazy tracks barely matter here: they do not extend the seek
range of a packet/EOF happens on them, they do not trigger seek range
joining, and they do not support backward demuxing.
This change should enable the following commit, while not causing any
behavior changes (i.e. bugs) with the current state.
A parameter that is actually used is removed from the param_names[]
array, so we can report unused parameters. This also happened on
duplicate parameters, so adjust the warning to make it less confusing.
(In any case, you're not supposed to provide duplicate parameters.)
Until now, delay-loading was for files with single tracks only
(basically what DASH and HLS like to expose, so adaptive streaming and
codec selection becomes easier - for sites, not for us). But they also
provide some interleaved versions, probably for compatibility. Until
now, we were forced to eagerly load it (making startup slightly slower).
But there is not much missing. We just need a way to provide multiple
metadata entries, and use them to represent each track.
A side effect is now that the "track_meta" header can be used for normal
EDL files too.
RTSP supports seeking, but at least the libavformat implementation makes
this dependent on runtime behavior. So you have to perform a seek, and
check if it fails. But even if you do this, the stream is interrupted
and restarted, and there seem to be other issues.
Assume that RTSP with unknown duration means it's a live stream, and
disable seeking in this case, as suggested by the issue reporter.
Fixes: #7472
Now this was stupid. To seek a source, it obviously has to be opened...
so just don't try to seek any unused source. If the track is actually
selected during playback, a seek to the correct position is performed
anyway.
These have ->segmented set (so the codec can be initialized properly),
but have no segment start or end times. This code was (probably) the
only thing which didn't handle this case.
ytdl_hook.lua can do this with all_formats and when delay_open is used,
and if the source stream actually contains both audio and video. In this
case, it might accidentally hide a media type completely, or waste
bandwidth (if the stream has true interleaved audio/video). So it's
important to warn.
While paused, the decoders typically stop reading data from the demuxer.
But for some reason, the file size is returned as a public field in
struct demuxer (wat...), and updated only when the packet reading
function is called. This caused the file size property to always return
the same value when paused, even though the demuxer thread was reading
new data, and the internal file size was updated.
Fix with a simple hack.
Instead of every packet. Doing it every packet led to the performance
regression mentioned in the fstat() commit. This should now be over, but
out of being careful, don't query the file size that often. This is only
used for user interface things, so this should not cause any problems.
For the sake of leaving the code compact, abuse another thing that is
updated only every second (speed statistics).
Libav seems rather dead: no release for 2 years, no new git commits in
master for almost a year (with one exception ~6 months ago). From what I
can tell, some developers resigned themselves to the horrifying idea to
post patches to ffmpeg-devel instead, while the rest of the developers
went on to greener pastures.
Libav was a better project than FFmpeg. Unfortunately, FFmpeg won,
because it managed to keep the name and website. Libav was pushed more
and more into obscurity: while there was initially a big push for Libav,
FFmpeg just remained "in place" and visible for most people. FFmpeg was
slowly draining all manpower and energy from Libav. A big part of this
was that FFmpeg stole code from Libav (regular merges of the entire
Libav git tree), making it some sort of Frankenstein mirror of Libav,
think decaying zombie with additional legs ("features") nailed to it.
"Stealing" surely is the wrong word; I'm just aping the language that
some of the FFmpeg members used to use. All that is in the past now, I'm
probably the only person left who is annoyed by this, and with this
commit I'm putting this decade long problem finally to an end. I just
thought I'd express my annoyance about this fucking shitshow one last
time.
The most intrusive change in this commit is the resample filter, which
originally used libavresample. Since the FFmpeg developer refused to
enable libavresample by default for drama reasons, and the API was
slightly different, so the filter used some big preprocessor mess to
make it compatible to libswresample. All that falls away now. The
simplification to the build system is also significant.
Add something that will access an URL embedded in EDL only when the
track it corresponds to is actually selected. This is meant to help with
ytdl_hook.lua and to improve loading speeds.
In theory, all this stuff is available to any mpv user, but discourage
using it, as it's so specialized towards ytdl_hook.lua, that there's
danger we'll just break this once ytdl_hook.lua stops using it, or
similar.
Mostly untested.
Normally, the first sub-stream is implicitly created. This change lets
the user use more orthogonal syntax, and use a new_stream header for
every sub-stream, instead of having to skip the header for the first
one.
Accidentally broken by commit 99700bc52c. mp_path_join() does not
check for this, because it's supposed to work on filesystem strings (and
e.g. "http://fubar" is a valid relative path in UNIX).
Add a mp_log context to the parse_edl() function, and report some
errors. Previously, this just told you that something was wrong. Add
some error reporting to make it slightly less evil.
Put all parameters in a list before processing them. This makes adding
parameters for special headers easier, and we can report parameters that
were not understood. (For "compatibility", which probably doesn't matter
at all, still don't count them as errors; as before.)
The timeline stuff has messed up memory management because there are no
clear ownership rules between a some demuxer instances (master or
demux_timeline) and the timeline object itself.
This is another subtle problem that happened: apparently,
demux_timeline.open is supposed to take over ownership of timeline, but
only on success. If it fails, it's not supposed to free it. It didn't
follow this, which lead to a double-free if demux_timeline.open failed.
The failure path in demux.c calls both timeline_destroy() and
demux_timeline.close on failure.
Move them around in the source code to get rid of the forward
declarations. Other than rearranging the lines and removing the 2
forward declarations, there are no other changes at all.
Apparently such .cue files exist. They fail both probing and parsing. To
make it worse, the sample at hand was encoded as Latin1.
One part of this is replacing bstr_lstrip() with a version that supports
NBSP. One could argue that bstr_lstrip() should always do this, but I
don't want to overdo it. There are many more unicode abomination which
it could be said it's supposed to handle, so it will stay ASCII instead
of going down this rabbit hole. I'm just assuming this cue sheet was
generated by some stupid software that inexplicably liked NBSPs (which
is how we justify a one-off fix). The new lstrip_whitespace() doesn't
look particularly efficient, but it doesn't have to be.
The second part is dealing with the fact that the charset is not
necessarily UTF-8. We don't want to do conversion before probing thinks
it knows it's a cue sheet (would probably make it more fragile all
around), so just make it work with Latin1 by assuming invalid code
points are Latin1. This fallback is part of why lstrip_whitespace() is
sort of roundabout.
(You could still rewrite it as much more efficient state machine,
instead of using a slow and validating UTF-8 parser that is called per
codepoint. Starting to overthink this.)
Multimedia is terrible. Legacy charsets are terrible. Everything is
terrible.
Fixes: #7429
See manpage additions. The libarchive behavior mentioned in the last
paragraph there is technically unrelated, but makes this new option
mostly pointless.
See: #7182
Until now, they were all just added to options.c (e.g. demux_mkv_conf).
This adds a mechanism which can be used to add future options in a
(very) slightly more elegant way.
Instead of opening every volume on start just to see if it's there, all
all volumes that could possibly exist, and "handle" it on opening. This
requires working around some of libarchive's amazing stupidity and using
some empirically determined behavior. Will possibly break if libarchive
changes some of this behavior.
See: #7182
When switching tracks, the data for the new track is missing by the
amount of data prefetched. This is because all demuxers return
interleaved data, and you can't just seek the switched track alone.
Normally, this would mean that the new track simply gets no data for a
while (e.g. silence if it's an audio track). To avoid this, mpv performs
a special "refresh seek" in the demuxer, which tries to resume demuxing
from an earlier position, in a way that does not disrupt decoding for
the non-changed tracks. (Could write a lot about the reasons for doing
something so relatively complex, and the alternatives and their
weaknesses, but let's not.)
This requires that the demuxer can tell whether a packet after a seek
was before or after a previously demuxed packet, sort of like an unique
ID. The code can use the byte position (pos) and the DTS for this. The
DTS is normally strictly monotonically increasing, the position in most
sane file formats too (notably not mp4, in theory).
The file at hand had DTS==NOPTS packets (which is fine, usually this
happens when PTS can be used instead, but the demux.c code structure
doesn't make it easy to use this), and pos==-1 at the same time. The
latter is what libavformat likes to return when the packet was produced
by a "parser" (or in other words, packets were split or reassembled),
and the packet has no real file position. That means the refresh seek
mechanism has no packet position and can't work.
Fix this by making up a pseudo-position if it's missing. This needs to
set the same value every time, which is why it does not work for
keyframe packets (which, by definition, could be a seek target).
Fixes: #7306 (sort of)
Although a linked list was ideal at first, there are cases where it
sucks, and became increasingly awkward (with the mpv command API
preferring integer indexes to access the list). In future, we probably
want to add more playlist-related functionality, so better change it to
an array now.
An array isn't always ideal either. Since playlist entries are still
separate objects (because in some cases you need a stable "iterator" to
it), but you still need to efficiently get the next/previous playlist
entry, there's a pl_index field, that needs to be maintained. E.g.
adding an entry at the start of the playlist => update the pl_index
field for all other entries. Well, it's not really worth to do something
more complicated to avoid these things.
This commit is probably buggy as shit. It's not like I bothered to test
everything. That's _your_ role.
Demuxers can call demux_close_stream() to close the underlying stream if
it's not needed anymore. (Useful to release "heavy" resources like FDs
and sockets. Plus merely keeping a file open can have visible side
effects such as inability to unmount a filesystem or, on Windows, to do
anything with the file.)
Until now, this set demuxer->stream to a dummy stream, because most code
used to assume that the stream field is non-NULL. But this requirement
disappeared (in some cases, the stream field is already NULL), so stop
doing that. demux_lavf.c, one of the demuxers which calls this function,
still had some of this, though.
mf:// has an obscure feature that lets you pass a list of filenames
separated by newlines. Who knows whether anyone is using that. It opened
these listfiles with fopen(), so the recent stream origin bullshit
doesn't operate on it. Fix this by using the mpv internal stream API
instead. Unfortunately there is no fgets(), so write an ad-hoc one. (An
implementation of line reading via "stream" is still in demux_playlist,
but it's better to keep it quarantined there.)
Until now, .edl files accepted only "simple" filenames, i.e. no relative
or absolute paths, no URLs. Now that the origin bullshit is a bit
cleaned up and enforced in the EDL code, there's absolutely no reason to
keep this.
The new code behaves somewhat similar to playlists. (Although playlists
are special because they're not truly recursively opened.)
mpv has a very weak and very annoying policy that determines whether a
playlist should be used or not. For example, if you play a remote
playlist, you usually don't want it to be able to read local filesystem
entries. (Although for a media player the impact is small I guess.)
It's weak and annoying as in that it does not prevent certain cases
which could be interpreted as bad in some cases, such as allowing
playlists on the local filesystem to reference remote URLs. It probably
barely makes sense, but we just want to exclude some other "definitely
not a good idea" things, all while playlists generally just work, so
whatever.
The policy is:
- from the command line anything is played
- local playlists can reference anything except "unsafe" streams
("unsafe" means special stream inputs like libavfilter graphs)
- remote playlists can reference only remote URLs
- things like "memory://" and archives are "transparent" to this
This commit does... something. It replaces the weird stream flags with a
slightly clearer "origin" value, which is now consequently passed down
and used everywhere. It fixes some deviations from the described policy.
I wanted to force archives to reference only content within them, but
this would probably have been more complicated (or required different
abstractions), and I'm too lazy to figure it out, so archives are now
"transparent" (playlists within archives behave the same outside).
There may be a lot of bugs in this.
This is unfortunately a very noisy commit because:
- every stream open call now needs to pass the origin
- so does every demuxer open call (=> params param. gets mandatory)
- most stream were changed to provide the "origin" value
- the origin value needed to be passed along in a lot of places
- I was too lazy to split the commit
Fixes: #7274
This has the advantage that playlists within the archive will work as
expected, because demux_playlist will correctly join the archive base
URL and entry name. Before this change, it could skip before the "|",
resulting in a broken URL.
EDL files can have multiple segments taken from the same source file. In
this case, the source file is supposed to be opened only once. This
stopped working, and it created a new demuxer instance for every single
segment entry. This made it slow and made it use much more memory than
needed.
This was because it tried to iterate over the array of source files, but
the array count (num_parts) was only set to a non-0 value later. Fix
this by maintaining the count correctly.
In addition, the actual code for checking whether a source can be reused
(in open_source()) regressed and stopped working correctly. d->stream
could be NULL. Use demuxer.filename instead; I'm not entirely sure
whether this is always correct, but fortunately we have a distributed
almost-AI driven test suite (called "users") which will probably find
and report such cases.
Probably broke with commit a09396ee60 or something close, but didn't
check closer.
Fixes: #7267
See previous commit. libavformat exports this information as AVStream.id
field.
The big problem is that the libavformat field is simply 0 if it's
unknown (i.e. the demuxer never sets it). So it needs to remain a
whitelist. Just add more formats which are known to have a meaningful
ID.
I considered exporting IDs for all formats, and then either leaving the
values as they are, or filtering duplicate values (and choosing
arbitrary but unique different IDs). But then again, I think it's sort
of mpv's job to filter FFmpeg's absurd bullshit API, and it should make
an effort to hide it rather than to reflect it.
See: #7211
The demuxer_id (exported in as "src-id" property) is supposed to be the
native stream ID, as it exists in the file, or -1 if that does not exist
(actually any negative value), or if it is unknown.
Until now, an ID was made up if it was missing. That seems like strange
non-sense, and I can't find the reason why it was done. But it was
probably for convenience by the EDL stuff or so.
Stop doing this. Fortunately, the src-id property was documented as
being unavailable if the ID is not known. Even the code for this was
present, it was just inactive until now. Extend input.rst with some
explanations.
Also fixing 3 other places where negative demuxer_id was ignored or
avoided.
Just an implementation detail that can be cleaned up now. Internally,
m_config maintains a tree of m_sub_options structs, except for the root
it was not defined explicitly. GLOBAL_CONFIG was a hack to get access to
it anyway. Define it explicitly instead.
If SEEK_FORWARD is set, a demuxer should skip to the next frame if the
timestamp does not fall on the start of a frame. If that flag is not
set, it should always seek to the first frame before the target
timestamp (or the first frame in the file).
It sometimes happens that HLS streams freeze because the HTTP server is
not responding for a fragment (or something similar, the exact
circumstances are unknown). The --timeout option didn't affect this,
because it's never set on HLS recursive connections (these download the
fragments, while the main connection likely nothing and just wastes a
TCP socket).
Apply an elaborate hack on top of an existing elaborate hack to somehow
get these options set. Of course this could still break easily, but hey,
it's ffmpeg, it can't not try to fuck you over. I'm so fucking sick of
ffmpeg's API bullshit, especially wrt. HLS.
Of course the change is sort of pointless. For HLS, GET requests should
just aggressively retried (because they're not "streamed", they're just
actual files on a CDN), while normal HTTP connections should probably
not be made this fragile (they could be streamed, i.e. they are backed
by some sort of real time encoder, and block if there is no data yet).
The 1 minute default timeout is too high to save playback if this
happens with HLS.
Vaguely related to #5793.
Until now, we've made FFmpeg use the default network timeout - which is
apparently infinite. I don't know if this was changed at some point,
although it seems likely, as I was sure there was a more useful default.
For most use cases, a smaller timeout is more useful (for example
recording something in the background), so force a timeout of 1 minute.
See: #5793
stream_skip() semantics were kind of bad, especially after the recent
change to the stream code. Forward stream_skip() calls could still
trigger a seek and fail, even if it was supposed to actually skip data.
(Maybe the idea that stream_skip() should try to seek is worthless in
the first place.)
Rename it to stream_seek_skip() (takes absolute position now because I
think that's better), and make it always skip if the stream is marked as
forward.
While we're at it, make EOF detection more robust. I guess s->eof
shouldn't exist at all, since it's valid only "sometimes". It should be
removed... but not today. A 1-byte stream_read_peek() call is good to
get the s->eof flag set to a correct value.
demux_mkv was the only thing using this, and everything else accessed it
directly. No need to keep the indirection wrapper around.
(Funny how this getter was in the initial commit of MPlayer.)
In some corner cases (see #6802), it can be beneficial to use a larger
stream buffer size. Use this as argument to rewrite everything for no
reason.
Turn stream.c itself into a ring buffer, with configurable size. The
latter would have been easily achievable with minimal changes, and the
ring buffer is the hard part. There is no reason to have a ring buffer
at all, except possibly if ffmpeg don't fix their awful mp4 demuxer, and
some subtle issues with demux_mkv.c wanting to seek back by small
offsets (the latter was handled with small stream_peek() calls, which
are unneeded now).
In addition, this turns small forward seeks into reads (where data is
simply skipped). Before this commit, only stream_skip() did this (which
also mean that stream_skip() simply calls stream_seek() now).
Replace all stream_peek() calls with something else (usually
stream_read_peek()). The function was a problem, because it returned a
pointer to the internal buffer, which is now a ring buffer with
wrapping. The new function just copies the data into a buffer, and in
some cases requires callers to dynamically allocate memory. (The most
common case, demux_lavf.c, required a separate buffer allocation anyway
due to FFmpeg "idiosyncrasies".) This is the bulk of the demuxer_*
changes.
I'm not happy with this. There still isn't a good reason why there
should be a ring buffer, that is complex, and most of the time just
wastes half of the available memory. Maybe another rewrite soon.
It also contains bugs; you're an alpha tester now.
The old code made it depend on ->seekable. If it isn't seekable, and
something discarded the data, then it'll just show an error message,
which will at least be somewhat informative. If no data was discarded,
the seek call is always a no-op.
There's a weird "timeline" condition in the old code; this doesn't
matter anymore, because timeline stuff does not pass streams down to
nested demuxers anymore.
demux_mkv has lots of logging that shows information about the file. It
sort of reminds of mkvinfo output. While this is sometimes interesting,
it's too much for verbose mode, and should be in debug log level.
This can be used by distros to disable all known FFmpeg ABI violations.
Currently only 1 is known, in demux_lavf.c. In addition to if-defing out
the access to the private FFmpeg field, this disables the possibly
fragile nested open callbacks, which make sense only if the
aforementioned field can be accessed.
In pseudo-DASH mode, we may have no real streams opened until the
demuxer layer is fully loaded and playback actually starts. The only
hint that the stream is from network is, at that point, the init
segment, which is only opened as stream, and then separately as demuxer
(which is dumb but happened to fit the internal architecture better).
So just propagate the flags from the init segment stream. Seems like an
annoyance, but doesn't hurt that much I guess. (Until someone gets the
idea to pass the init segment data inline or so, but nothing does that.)
The sample link in the linked issue will probably soon switch to another
format, because that service always does this after recent uploads or
so.
Fixes: #7038
Someone crazy is trying to mix images with videos in EDL files. Putting
an image as first thing into the EDL disabled audio, because the first
EDL entry was used to define the layout.
Change this. Make it user-configurable, and use a "better" heuristic to
select the default otherwise.
In theory, EDL could be easily extended to specify track layout and
mapping of parts to virtual EDL tracks manually and in great detail. But
I don't think it's worth it - who would bother using it?
Fixes: #6764
This partially reverts commit a9d83eac40
("Remove optical disc fancification layers").
Mostly due to the timestamp crap, this was never really going to work.
The playback layer is sensitive to timestamps, and derives the playback
time directly from the low level packet timestamps. DVD/BD works
differently, and libdvdnav/libbluray do not make it easy at all to
compensate for this. Which is why it never worked well, but not doing it
at all is even more awful.
demux_disc.c tried this and rewrote packet timestamps from low level TS
to playback time. So restore demux_disc.c, which should bring behavior
back to the old often non-working but slightly better state.
I did not revert anything that affects components above the demuxer
layer. For example, the properties for switching DVD angles or listing
disc titles are still gone. (Disc titles could be reimplemented as
editions. But not by me.)
This commit modifies the reverted code a bit; this can't be avoided,
because the internal API changed quite a bit. The old seek resync in
demux_lavf.c (which was a hack) is replaced with a hack. SEEK_FORCE and
demux_params.external_stream are new additions.
Some of this could/should be further cleaned up. If you don't want
"proper" DVD/BD support to disappear, you should probably volunteer.
Now why am I wasting my time for this? Just because some idiot users are
too lazy to rip their ever-wearing out shitty physical discs? Then why
should I not be lazy and drop support completely? They won't even be
thankful for me maintaining this horrible garbage for no compensation.
This was added in 585f9ff42f by @bbarenblat (github handle). We
don't do this. This file alone probably has multiple dozen of authors (I
didn't count, but it has a history of 15 years). If everyone added their
names with each small change, this project would have giant lists of
contributing authors on every source file.
Neither copyright law nor any of the used licenses require listing
authors in the license header. Authorship is recorded in the git log.
So don't start with this, and remove this recent case to avoid setting a
precedent.
Some files still have an author in the header. These cases are
grandfathered, and usually are the actual authors of the original code.
demux_mkv may seek to the end of the file to read certain headers (which
should probably be called "footers", but in theory they are just headers
that have been placed at the end of the file unfortunately).
This commit changes behavior not to seek if the stream is not marked as
seekable. Before this, it only checked whether the stream size was
unknown (end negative). In practice it doesn't make much of a
difference, since seekable usually equals known stream size.
Also improve the wording, and distinguish between actual incomplete
files, and unseekable ones.
This passed all streams to mp_recorder_create(), even disabled ones. The
disabled streams never get packets, so recorder.c eventually errors out
with unrelated-looking errors. The reason is that recorder.c waits for
packets to appear on other streams, which in turn is because libavformat
refuses to mux empty streams anyway.
recorder.c could call demux_stream_is_selected(), which would have made
the patch much smaller. But this feels like a bad idea, since recorder.c
should use sh_stream only for metadata (and not in an "active" way), nor
should it care what demux.c is currently doing with it. So make the API
user (demux.c) pass only the streams it really wants.
Fixes: #6999
Although this was sort of elegant, it just seems to complicate things
slightly. Originally, the API meant that you cache mp_recorder_sink
yourself (which would avoid the mess of passing an index around), but
that too seems slightly roundabout.
In a later change, I want to change the set of streams passed to
mp_recorder_create(), and then I'd have to keep track of the index for
each stream, which would suck. With this commit, I can just pass the
unambiguous sh_stream to it, and it will be guaranteed to match the
correct stream.
The disadvantages are barely worth discussing. It's a new linear search
per packet, but usually only 2 to 4 streams are active at a time. Also,
in theory a user could want to write 2 streams using the same sh_stream
(same metadata, just writing different packets or so), but in practice
this is never done.
in->eof is used as an indicator whether reading packets still makes
sense. (Without this, the prefetcher would obviously burn CPU by
retrying reading even though everything has been read.)
This was not reset properly after seeks were performed. It led to
getting stuck in at least one corner case: when enabling a track, the
demuxer would seek backwards to get new packets from the current
playback position ("refresh seeks"). But if playback was paused, and EOF
was previously reached, it would not try to read packers again due to
in->eof being false. There was not anything else that would make it
retry reading, so it was stuck in a weird underrun/buffering state.
Fixes: #6986
This was a leftover from commit b2752321 which fixed#6522 but after
the recent demux refactoring this fix is superseded by commit 0f6cda4ab.
Remove the redundant update call.
This detected the first packet demuxed after a seek as timestamp
discontinuity. Obviously this is non-sense. Since the OGG radio streams
for which this feature was introduced are normally unseekable, it's
simple to fix this: simply disable it (if in auto mode, the default) as
soon as a seek is performed. This code is never called if the stream is
considered unseekable, unless the user forced it.
There's still a chance this linearization is performed before a seek
happens. This will be a bit awkward, but no worse than without this
feature, since seeking with timestamp resets is inherently broken in
both mpv and libavformat.
Fixes: #6974
Fixes: 27fcd4d
Before this commit, EDL or CUE files did not properly enable the cache
if they were on "slow" media (stream->streaming==true). This happened
because the stream is unset for demux_timeline, so the streaming flag
could not be queried anymore.
Fix this by adding this flag to struct demuxer, and propagate it exactly
like the is_network flag. is_network is not used for checking the cache
options anymore, and its main function seems to be something else.
Normal http streams set the streaming flag already.
This should fix#6958.
This field is documented as internal, so an API user should not
access it. However, this is the only way to get some read statistics
without replacing FFmpeg's entire HLS demuxer. (Using custom I/O as
workaround doesn't work: the HLS code uses some weird internal APIs
that cannot be provided by FFmpeg API users; I even made the author
of the relevant patch to provide a public API, but which was shot
down by another FFmpeg developer. So I take this as my right to
access this field.)
Mention this explicitly, as it affects ABI and API compatibility, and
I don't want that anyone claims this was a "mistake". Add some
explanations.
In theory, a 0 size allocation could have made it memset() on a NULL
pointer (with a non-0 size, which makes it crash in addition to
theoretical UB).
This should never happen, since even packets with size 0 should have an
associated allocation, as FFmpeg currently does. But avoiding this makes
the API slightly more orthogonal and less tricky, I guess.
This reverts commit 95636c65e7.
This change shouldn't be needed, and in fact it's wrong. The FFmpeg API
function could do anything it wants with the packet, including changing
the packet data pointer. Likewise, it's not guaranteed that the
referenced packet's fields mirror the current state of the mpv packet
struct (the AVPacket is only kept for the AVBuffer and the side data
stuff).
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.
The step_backwards function set reader_head to the start of the current
cache range. This was completely unnecessary and made it _much_ slower.
Remove the code that adjusts reader_head. Merge the rest of the code
into the only caller and remove the function.
The comment on the removed code was quite right. It was "inefficient".
Removing it delegates going to an early position to the normal seek
code, triggered by find_backward_restart_pos() incremental back seek
logic. I suppose especially audio benefits from this, because this
happens for every single audio packet (except maybe freaky bullshit like
TrueHD, which has "keyframes").
The blabla about performance in the removed comments is still true, but
now applies to the seek code itself only.
Fixes "mpv file.mkv --cache --demuxer-cache-wait --play-dir=backward",
and other situations where the demuxer cache contains the entire file,
and playback is to start from the end. It also can be triggered when
starting playback normally with --cache, and once everything is in the
cache, enabling backward playback and seeking past EOF.
In all cases, the cache seek will set reader_head=NULL (because you
seeked to/past EOF). Then the code (the one modified by this commit)
sees that ds->queue->is_bof==true, and thinks we've reached BOF
(beginning of file) while searching for a useful packet, i.e. we found
nothing and playback really can only end.
Obviously this is nonsense, we've found only nothing if we actually
searched from the beginning, not some "random" reader_head (== first)
value that does not include the entire cache. That means the condition
should trigger only if the start of the search (first variable) points
to the beginning of the cache (ds->queue->head).
Not taking this if means we'll seek to an earlier position and retry.
Also, a seek before the beginning of the cache will always end up with
reader_head==ds->queue->head, i.e. we'll terminate properly.
That comment was quite right.
Together with the previous commit, this seems to make backward playback
work in files with vorbis and mp3 audio codecs.
For Vorbis (with libavcodec's decoder, didn't test libvorbis), the first
packet was just always completely discarded. This happened even though
we tell libavcodec that we do discarding of padding manually. It simply
happened inside the codec, not libavcodec's general initial padding
handling. In addition, the first output decoded frame seems to contain
partial data. (Unlike the opus decoder, it doesn't report any padding at
all.)
The Opus decoder (again libavcodec only tested) reports an initial
padding, but it appears to be too small, and it sounds right only with 2
packets discarded. So its status doesn't change.
I'm not sure why I need 2 frames for mp3, but with that value I had
success on the samples I tested.
Matroska has this weird concept of "lacing", which are really sub-blocks
packed into a larger actual block. They are demuxed as individual
packets, because that's what the decoder needs. Basically they're a
Matroska hack to reduce per-packet muxing overhead.
One problem is that these sub-blocks don't have timestamps. The
timestamps need to be created from the "default duration". If this
default duration isn't in the file header (or if we drop it when it has
a known broken value), the resulting packets won't have a timestamp.
This is an usual and weird situation, that may confuse the demuxer layer
(and backward playback in particular). Fix this by not setting the
keyframe flag for these.
This affects only audio packets. Subtitle lacing is explicitly not
supported (in theory would at least lack a way to specify durations),
and video won't usually use it for real video codecs (not every frame is
a "keyframe", timestamp reordering).
Shitty ancient hack that wastes my time all the time.
demux.c: always return the coverart packet as soon as possible, and
don't let the backward demux state machine possibly stop it.
f_decoder_wrapper.c: mess with some shit until it somehow starts to
work. I think the old code tried to let it cleverly fall through so the
packet was processed "normally"; just make it run the "usual" code
instead.
This commit generally fixes backward playing in wav, at least in most
PCM cases.
libavformat's wav demuxer (and actually all other raw PCM based
demuxers) have a specific behavior that breaks backward demuxing. The
same thing also breaks persistent seek ranges in the demuxer cache,
although that's less critical (it just means some cached data gets
discarded). The backward demuxing issue is fatal, will log the message
"Demuxer not cooperating.", and then typically stop doing anything.
Unlike modern media formats, these formats don't organize media data in
packets, but just wrap a monolithic byte stream that is described by a
header. This is good enough for PCM, which uses fixed frames (a single
sample for all audio channels), and for which it would be too expensive
to have per frame headers.
libavformat (and mpv) is heavily packet based, and using a single packet
for each PCM frame causes too much overhead. So they typically "bundle"
multiple frames into a single packet. This packet size is obviously
arbitrary, and in libavformat's case hardcoded in its source code.
The problem is that seeking doesn't respect this arbitrary packet
boundary. Seeking is sample accurate. You can essentially seek inside a
packet. The resulting packets will not be aligned with previously
demuxed packets. This is normally OK.
Backward seeking (and some other demuxer layer features) expect that
demuxing an earlier demuxed file position eventually results in the same
packets, regardless of the seeks that were done to get there. I like to
call this "deterministic" demuxing. Backward demuxing in particular
requires this to avoid overlaps, which would make it rather hard to get
continuous output.
Fix this issue by detecting wav and hopefully other raw audio formats
with a heuristic (even PCM needs to be detected as heuristic). Then, if
a seek is requested, align the seek timestamps on the guessed number of
samples in the audio packets returned by the demuxer.
The heuristic excludes files with multiple streams. (Except "attachment"
video streams, which could be an ID3 tag. Yes, FFmpeg allows ID3 tags on
WAV files.) Such files will inherently use the packet concept in some
way.
We don't know how the demuxer chooses the internal packet size, but we
assume that it's fixed and aligned to PCM frame sizes. The frame size is
most likely given by block_align (the native wav frame size, according
to Microsoft). We possibly need to explicitly read and discard a packet
if the seek is done without reading anything before that. We ignore any
subsequent packet sizes; we need to avoid the very last packet, which
likely has a different size.
This hack should be rather benign. In the worst case, it will "round"
the seek target a little, but the maximum rounding amount is bounded.
Maybe we _could_ round up if SEEK_FORWARD is specified, but I didn't
bother.
An earlier commit fixed the same issue for mpv's demux_raw.
An alternative, and probably much better solution would be clipping
decoded data by timestamp. demux.c could allow the type of overlap the
wav demuxer introduces, and instruct the decoder to clip the output
against the last decoded timestamp. There's already an infrastructure
for this (demux_packet.end field) used by EDL/ordered chapters.
Although this sounds like a good solution, mpv unfortunately uses floats
for timestamps. The rounding errors break sample accuracy. Even if you
used integers, you'd need a timebase that is sample accurate (not always
easy, since EDL can merge tracks with different sample rates).
Yay, more subtle state on top of this nightmarish, fragile state
machine. But this is what happens when you subvert the laws of nature.
This simple checks where playback should "resume" from when no packets
were returned to the decoder yet after the seek that initiated backward
playback. The main purpose is to process the first returned keyframe
range in the same way like all other ranges. This ensures that things
like preroll are included properly.
Before this commit, it could for example have happened that the start of
the first audio frame was slightly broken, because no preroll was
included. Since the audio frame is reversed before sending it to the
audio output, it would have added an audible discontinuity before the
second frame was played; all subsequent frames would have been fine.
(Although I didn't test and confirm this particular issue.)
In future, this could be useful for certain other things.
At least the condition for delaying the backstep seek becomes simpler
and more explicit.
Move the code that attempts to start demuxing up in dequeue_packet.
Before, it was not called when the stream was in back_restarting state.
This commit makes streams be in back_restarting state at initialization,
so the demuxer would never have started reading.
Likewise, we need to call back_demux_see_packets() right after seek in
case the seek was within the cache. (We don't bother with checking
whether it was a cached seek; nothing happens if it was a normal one.)
There is nothing else that would process these cached packets
explicitly, although coincidences could sporadically trigger it.
The check for back_restart_next in find_backward_restart_pos() now
decides whether to use this EOF special code. Since the backward
playback start state also sets this variable, we don't need some of
the complex checks in dequeue_packet() anymore either.
Make --audio-backward-overlap default to 2 for Opus. I have no idea why
this is needed. It seems to fix backward decoding though (going purely
by listening).
Normally, this should not be needed, since initial padding is completely
contained within the first packet (normally, and in the case I tested).
So the 2nd packet/frame should be fine, but for some unknown reason it
works only with the 3rd.
This seems more useful in general. This change also happens to fix a
miscounting of preroll packets when some of them were "rounded" away,
and which could make it stuck.
Also a simple intra-refresh encode with x264 (and muxed to mkv by it)
seems to work now. I guess I misinterpreted earlier results.
Backstepping still could get "stuck" if the demuxer didn't seek far back
enough. This commit fixes getting stuck if playing backwards from the
end, and audio has ended much earlier than the video.
In commit "demux: fix initial backward demuxing state in some cases",
I claimed that the backward seek semantics ("snapping" backward in
normal seeking, unrelated to backward playing) would take care of
this. Unfortunately, this is not always quite true.
In theory, a seek to any position (that does not use SEEK_FORWARD, i.e.
backward snapping) should return a packet for every stream. But I have a
mkv sample, where audio ends much earlier than video. Its mkvmerge
created index does not have entries for audio packets, so the video
index is used. This index has its last entry somewhere close after the
end of audio. So no audio packets will be returned. With a "too small"
back_seek_size, the demuxer will retry a seek target that ends up in
this place forever. (This does not happen if you use --index=recreate.
It also doesn't happen with libavformat, which always prefers its own
index, while mpv's internal mkv demuxer strictly prefers the index from
the file if it can be read.)
Fix this by adding the back_seek_size every time we fail to see enough
packets. This way the seek step can add up until it works.
To prevent that back_seek_pos just "runs away" towards negative infinity
by subtracting back_seek_size every time we back step to undo forward
reading (e.g. if --no-cache is used), readjust the back_seek_pos to the
lowest known resume position. (If the cache is active, kf_seek_pts can
be used, but to work in all situations, the code needs to grab the
minimum PTS in the keyframe range.)
Just rearranging shit. Setting SEEK_HR for backstep seeks actually
doesn't have much meaning, but disables the weird audio snapping for
"keyframe" seeks, and I don't know it's late.
This code used to be simpler, but now it's enough that it should be
factored into a single function.
Both uses of the new function are annoyingly different. The first use is
the special case when a decoder tries to read packets, but the demuxer
doesn't see any (like mp4 files with sparse video packets, which
actually turned out to be chapter thumbnail "tracks"). Then the other
stream queues will overflow, and the stream with no packets is marked
EOF to avoid stalling playback.
The second case is when the demxuer returns global EOF.
It would be more awkward to have the loop iterating the streams in the
function, because then you'd need a weird parameter to control the
behavior.
Just "mpv file.mkv --play-direction=backward" did not work, because
backward demuxing from the very end was not implemented. This is another
corner case, because the resume mechanism so far requires a packet
"position" (dts or pos) as reference. Now "EOF" is another possible
reference.
Also, the backstep mechanism could cause streams to find different
playback start positions, basically leading to random playback start
(instead of what you specified with --start). This happens only if
backstep seeks are involved (i.e. no cached data yet), but since this is
usually the case at playback start, it always happened. It was racy too,
because it depended on the order the decoders on other threads requested
new data. The comment below "resume_earlier" has some more blabla.
Some other details are changed.
I'm giving up on the "from_cache" parameter, and don't try to detect the
situation when the demuxer does not seek properly. Instead, always seek
back, hopefully some more.
Instead of trying to adjust the backstep seek target by a random value
of 1.0 seconds. Instead, always rely on the random value provided by the
user via --demuxer-backward-playback-step. If the demuxer should really
get "stuck" and somehow miss the seek target badly, or the user sets the
option value to 0, then the demuxer will not make any progress and just
eat CPU. (Although due to backward seek semantics used for backstep
seeks, even a very small seek step size will work. Just not 0.)
It seems this also fixes backstepping correctly when the initial seek
ended at the last keyframe range. (The explanation above was about the
case when it ends at EOF. These two cases are different. In the former,
you just need to step to the previous keyframe range, which was broken
because it didn't always react correctly to reaching EOF. In the latter,
you need to do a separate search for the last keyframe.)
Fixes the same thing as the previous commit did with demux_mkv. I'm not
sure if this is correct or a good idea (well, it works with my sample
file).
There are some shady things in this, but describing them would require
too many expletives.
In this scenario, the demuxer will output timestamps offset by the codec
delay (e.g. negative timestamps at the start; mkv simulates those), and
the trimming in the decoder (often libavcodec, but ad_lavc.c in our
case) will adjust the timestamps back (e.g. stream actually starts at
0).
This offset needs to be taken into account when seeking. This worked in
the uncached case. (demux_mkv.c is a bit tricky in that the index is
already in the offset space, so it compensates even though the seek call
does not reference codec_delay.) But in the cached case, seeks backwards
did not seek enough, and forward they seeked too much.
Fix this by adding the codec delay to the index search. We need to get
"earlier" packets, so e.g. seeking to position 0 really gets the initial
packets with negative timestamps.
This also adjusts the seek range start. This is also pretty obvious: if
the beginning of the file is cached, the seek range should start at 0,
not a negative value. We compare 0-based timestamps to it later on.
Not sure if this is the best approach. I also could have thought
about/checked some corner cases harder. But fuck this shit.
Not fixing duration (who cares) or end trimming, which would reduce the
seek range and duration (who cares).
This is a bad approach, and should be handled by a codec parameter field
(in mp_codec_params or AVCodecParameters).
It's bad because it's overly complicated, and has potential to break
demuxer cache assumptions: packets that were "intended" for seek
resuming may suddenly appear in the middle of a stream, when you seek
back and play a cached part again. (In general it was fine though,
because seek range joining tends to remove the first audio packet of the
next range when trying to find an overlap.)
demux_mkv.c does not try to export its codec_delay field through the
codec parameters mentioned above. In the only case I spotted this
element, the codec itself (opus) set this field within libavcodec. And I
think that's actually how it should be. On the other hand, a file could
in theory set this field via mkv headers if the codec is too stupid to
have such a field internally. But I don't really care until I see such a
file.
The end trimming is still sort of needed (though not sure if anything
uses it, other than the opus/mkv test sample I was using). The decoder
can't know whether something is the last packet, until it's too late.
The codec_delay field is still needed to offset timestamps.
Only timestamps that enter or leave the demuxer API should be adjusted
by ts_offset (which is usually the start time). queue_seek() is also
used by backward demux seeks, which uses an internal timestamp.
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.
This is for uncompressed data, so every frame is a "keyframe". This is
part of making this demuxer work with the demuxer layer caching and
backward playback.
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.
The demuxer layer can start a thread to decouple the rest of the player
from blocking I/O (such as network accesses). But this particular
function does not support running with the thread enabled. The mutex use
within it is only since thread_work() may temporarily unlock the mutex,
and unlocking an unlocked mutex is not allowed. Most of the rest of the
code still does proper locking, even if it's pointless and effectively
single-threaded.
To make this look slightly cleaner, extend the mutex around the rest of
the code (like threaded code would have to do). This is mostly a
cosmetic change.
The demuxer cache benefits slightly from knowing where the current file
or stream begins. For example, seeking "left most" when the start is
cached would not trigger a low level seek (which would be followed by
messy range joining when it notices that the newly demuxed packets
overlap with an existing range).
Unfortunately, since multimedia is so crazy (or actually FFmpeg in its
quite imperfect attempt to be able to demux anything), it's hard to tell
where a file starts. There is no feedback whether a specific seek went
to the start of the file. Packets are not tagged with a flag indicating
they were demuxed from the start position. There is no index available
that could be used to cross-check this (even if the file contains a full
and "perfect" index, like mp4). You could go by the timestamps, but who
says streams start at 0? Streams can start somewhere at an extremely
high timestamps (transport streams like to do that), or they could start
at negative times (e.g. files with audio pre-padding will do that), and
maybe some file formats simply allow negative timestamps and could start
at any negative time. Even if the affected file formats don't allow it
in theory, they may in practice. In addition, FFmpeg exports a
start_time field, which may or may not be useful. (mpv's internal mkv
demuxer also exports such a field, but doesn't bother to set it for
efficiency and robustness reasons.)
Anyway, this is all a huge load of crap, so I decided that if the user
performs a seek command to time 0 or earlier, we consider the first
packet demuxed from each stream to be at the start of the file. In
addition, just trust the start_time field. This is the "shitty" part of
this commit.
One common case of negative timestamps is audio pre-padding. Demuxers
normally behave sanely, and will treat 0 as the start of the file, and
the first packets demuxed will have negative timestamps (since they
contain data to discard), which doesn't break our assumptions in this
commit. (Although, unfortunately, do break some other demuxer cache
assumptions, and the first cached range will be shown as starting at a
negative time.)
Implementation-wise, this is quite simple. Just split the existing
initial_state flag into two, since we want to deal with two separate
aspects. In addition, this avoids the refresh seek on track switching
when it happens right after a seek, instead of only after opening the
demuxer.
There were 3 packet reading functions: the "old" demux_read_packet()
that blocked (leftover from MPlayer times, but was still used until
recently by some obscure code), the "new" demux_read_packet_async(), and
the special demux_read_any_packet(), that is used by pseudo-demuxers
like demux_edl.
The first two could be used both in threaded and un-threaded mode. This
made 5 cases in total. Some bits of logic was spread across all of them.
Unify the logic. A recent commit made demux_read_packet() private, and
the code for it in threaded mode disappears. The difference between
threaded and un-threaded is minimized.
It's possible that this commit causes random regression. Enjoy.
Shreesh Adiga