| Commit message (Collapse) | Author | Age | Files | Lines |
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There isn't really a need to disable this for local playback. I think
originally I did this because I was afraid the code could mess up or be
annoying on local mode, but that's not really a good argument. I'd
rather test this code in local mode too. In this case, it shouldn't
really happen that it runs out of cache in the first place.
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The --cache-pause feature (enabled by default) will pause playback for a
while if network runs out of data. If this is not done, then playback
will go on frame-wise (as packets are slowly read from the network and
then instantly decoded and displayed). This feature is actually useless,
as you won't get nice playback no matter what if network is too slow,
but I guess I still prefer this behavior for some reason.
This commit changes this behavior from using the demuxer cache state
only, to trying to use underrun information from the AO/VO. This means
if you have a very large audio buffer, then cache-pausing will trigger
once that buffer is depleted, which will be some time _after_ the
demuxer cache has run out.
This requires explicit support from the AO. Otherwise, the behavior
should be mostly the same as before this commit.
This does not care about the AO buffer. In theory, the AO may underrun,
then the player will write some data to the AO buffer, then the AO will
recover and play this bit of data, then the player will probably trigger
the cache-pause behavior. The probability of this happening should be
pretty low, so I will hold off fixing this until the next refactor of
the AO chain (if ever).
The VO underflow detection was devised and tested in 5 minutes, and may
not be correct. At least I'm fairly sure that the combination of all the
factors should make incorrect behavior relatively unlikely, but problems
are possible.
Also, the demux_reader_state.underrun field may be inaccurate. It's only
the present state at the time demux_get_reader_state() was called, and
may exclude past underruns. In theory, this could cause "close" cases to
be missed. Then you might get an audio underrun without cache-pausing
acting on it. If the stars align, this could happen multiple times in
the row, effectively making this feature not work.
The most user-visible consequence of this change is that the user
will now see an AO underrun warning every time the cache runs out.
Maybe this cache-pause feature should just be removed...
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If a file format supports PTS resets, get_current_pos_ratio calculates
the ratio using the stored filepos in the demuxer struct instead of a
standard query of the current time in the stream and its total length.
This seems like a reasonable way to avoid weird PTS values, but in
reality this just causes problems and results in inaccurate ratio
values that can affect other parts of the player (most notably the osc
seekbar).
For libavformat formats, demux->filepos is obtained from the
demux_lavf_fill_buffer function which is called on the next packet. The
problem is that there is a slight delay between packets and in some
cases, this delay can be relatively large. That means the obtained
demuxer->filepos value will be very inaccurate since it obtains the pos
from the end of the upcoming packet and not its actual current position.
This is especially noticeable at the very beginning of playback where
get_current_pos_ratio sometimes returns a value of well over 2% despite
less than a second passing in the stream. Another telltale sign is to
simply watch the osc seekbar as a stream progresses and observe how it
loads in staggered steps as every packet is decoded. In contrast, the
seekbar progresses smoothly on the playback of a format that does not
support PTS resets. The simple solution is to instead use the query of
the current time and length of a stream and calculate the ratio that
way.
get_current_pos_ratio will still fallback on using the byte stream
position if the previous queries fail. However, get_current_time will
be more accurate in the vast majority of cases and should be the
preferred method of calculating the position ratio.
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Track switching doesn't run reset_playback_state(), so a track enabled
at runtime during backward playback would lead to a messed up state.
This commit just does a bad code monkey fix to this. It feels like there
needs to be a much better way to propagate this state.
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Another shitty obscure feature that usually nobody notices.
Unsurprisingly, it doesn't go well with backward playback mode.
If you use --keep-open in forward playback mode, and seek past the end
of the file, it tries to seek to the very last frame. The demuxer will
seek to the last "keyframe" before the end (i.e. some frames to go in
most cases), and trying to hr-seek to the file duration often won't cut
it, so this requires some special code. The function at hand seeks
"close" to the end, and then stops hr-seek when the last frame us
encountered (simple enough and very effective).
In backward playback mode, start and end are reversed, and we need to
seek "close" to the start of the file instead. Simple enough to do, and
it works.
One problem is that command.c has some weird logic to make going beyond
the last chapter either end playback (--keep-open=no), or jump to the
last frame. Now this will jump to the first frame, which is weird, but
let's ignore this.
Another problem is that seeking before playback start position hits EOF
in backward playback mode, which is a demuxer bug, and has nothing to do
with this code. But it triggers this code, so seeking before the start
will show the "last" frame. (My description is a mess with directions.
Figure it out yourself.)
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Obviously should seek back to the end of the file when it loops.
Also remove some minor code duplication around start times. This isn't
the correct solution by the way. Rather than hoping we know a reasonable
start/end time, this stuff should instruct the demuxer to seek to the
exact location. It'll work with 99% of all normal files, but add an
appropriate comment (that basically says the function is bullshit) to
get_start_time() anyway.
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Not sure if this is better or worse.
Some minor behavior changes.
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This changes the behavior of the --ab-loop-a/b options. In addition, it
makes it work with backward playback mode.
The most obvious change is that the both the A and B point need to be
set now before any looping happens. Unlike before, unset points don't
implicitly use the start or end of the file. I think the old behavior
was a feature that was explicitly added/wanted. Well, it's gone now.
This is because of 2 reasons:
1. I never liked this feature, and it always got in my way (as user).
2. It's inherently annoying with backward playback mode.
In backward playback mode, the user wants to set A/B in the wrong order.
The ab-loop command will first set A, then B, so if you use this command
during backward playback, A will be set to a higher timestamps than B.
If you switch back to forward playback mode, the loop would stop
working. I want the loop to just continue to work, and the chosen
solution conflicts with the removed feature.
The order issue above _could_ be fixed by also switching the AB-loop
user option values around on direction switch. But there are no other
instances of option changes magically affecting other options, and doing
this would probably lead to unexpected misery (dying from corner cases
and such).
Another solution is sorting the A/B points by timestamps after copying
them from the user options. Then A/B options set in backward mode will
work in forward mode. This is the chosen solution. If you sort the
points, you don't know anymore whether the unset point is supposed to
signify the end or the start of the file.
The AB-loop code is slightly better abstracted now, so it should be easy
to restore the removed feature. It would still require coming up with a
solution for backwards playback, though.
A minor change is that if one point is set and the other is unset, I'm
rendering both the chapter markers and the marker for the set point.
Why? I don't know. My test file had chapters, and I guess I decided this
looked better.
This commit also fixes some subtle and obvious issues that I already
forgot about when I wrote this commit message. It cleans up some minor
code duplication and nonsense too.
Regarding backward playback, the code uses an unsanitary mix of internal
("transformed") and user timestamps. So the play_dir variable appears
more than usual.
To mention one unfixed issue: if you set an AB-loop that is completely
past the end of the file, it will get stuck in an infinite seeking loop
once playback reaches the end of the file. Fixing this reliably seemed
annoying, so the fix is "just don't do this". It's not a hard freeze
anyway.
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This code attempts to seek to the last frame by seeking close to the
end, and then decoding until the last frame has been reached. To do so
it sets hrseek_lastframe, which for video enables some logic to "catch"
this last frame, and completely ignores hrseek_pts. But audio still may
use hrseek_pts
I don't know if the original author (me) was thinking, if anything, when
setting this variable to 1e99, essentially a random, number. It's very
large, and a timestamp like this will never happen, so it does its job.
But it's random.
Use INFINITY instead. It will skip all audio samples in the audio code
correctly. This change doesn't fix anything, but it does get rid of the
random looking number.
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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.
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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.
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This will properly notify observed properties if the player hasn't
started actual playback yet, such as with --demuxer-cache-wait.
This also happens to cause the main loop more often, which triggers
MPV_EVENT_IDLE, and fixes the OSC display. (See previous commit
message.)
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The OSC's (osc.lua) event handling is fundamentally broken. It waits for
MPV_EVENT_TICK to update the UI, and MPV_EVENT_TICK has become entirely
meaningless, except as a hack for the OSC. There are many situations
where the OSC doesn't properly update because the TICK event it expects
isn't sent.
Fix one of them: it doesn't update the cache state if the VO window is
forced and --demuxer-cache-wait is used. Make it so that the tick event
is sent even if playback initialization is taking time.
This is still slightly broken, because it works only if the mainloop is
actually run, which depends on random circumstances (such as moving the
mouse over the VO window). The next commit will add another such
circumstance which will make it appear to work, although it's still
conceptually broken. If we "fixed" it and strictly woke up the player
if the idle timer ran out, we'd send tick events all the time, even
if nothing is going on, which we don't want. Fucking shitshow.
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Instead of going through those weird DEMUXER_CTRLs, query this
information directly. I'm not sure which kind of brain damage made me
use CTRLs for these. Since there are no other DEMUXER_CTRLs that make
sense for the frontend, remove the remaining infrastructure for them
too.
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The stream size return was the only thing that still required doing
STREAM_CTRLs from frontend through the demuxer layer. This can be done
much easier, so rip it out. Also rip out the now unused infrastructure
for STREAM_CTRLs via demuxer layer.
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This prevents the pause state from triggering before the audio output is
finished playing back audio. This is particularly helpful for gapless
audio.
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Signed-off-by: Aman Gupta <aman@tmm1.net>
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Adds the negation missed in 8816e1117ee65039dbb5700219ba3537d3e5290e
when moving from a positive-is-active to positive-is-idle variable.
This leads to proper updates to properties such as "eof-reached",
as well as fixes screensaver state updates.
Separately found and fixed by avih and wnoun.
Co-authored-by: wnoun <wnoun@outlook.com>
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The demuxer cache is the only cache now. Might need another change to
combat seeking failures in mp4 etc. The only bad thing is the loss of
cache-speed, which was sort of nice to have.
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I'm also not sure whether this condition doesn't subtly break a lot of
things.
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Before this, mpctx->playing was often used to determine whether certain
new state could be added to the playback state. In particular this
affected external files (which added tracks and demuxers). The variable
was checked to prevent that they were added before the corresponding
uninit code. We want to make a small part of uninit asynchronous, but
mpctx->playing needs to stay in the place where it is. It can't be used
for this purpose anymore.
Use mpctx->stop_play instead. Make it never have the value 0 outside of
loading/playback. On unloading, it obviously has to be non-0.
Change some other code in playloop.c to use this, because it seems
slightly more correct. But mostly this is preparation for the following
commit.
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This is nonsense. Didn't matter in most situations, because seeking
itself set this after it was cleared. But some callers don't do this,
see e.g. commit ed73ba89644fc6. There is no need to clear it at all, and
it causes issues with the next commit. It only needs to be reset on
loading.
Also move the initialization on loading up, which doesn't change
behavior, but makes the intention clearer.
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Many asynchronous commands are potentially long running operations, such
as loading something from network or running a foreign process.
Obviously it shouldn't just be possible for them to freeze the player if
they don't terminate as expected. Also, there will be situations where
you want to explicitly stop some of those operations explicitly. So add
an infrastructure for this.
Commands have to support this explicitly. The next commit uses this to
actually add support to a command.
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This enables two types of command behavior:
1. Plain async behavior, like "loadfile" not completing until the file
is fully loaded.
2. Running parts of the command on worker threads, e.g. for I/O, such as
"sub-add" doing network accesses on a thread while the core
continues.
Both have no implementation yet, and most new code is actually inactive.
The plan is to implement a number of useful cases in the following
commits.
The most tricky part is handling internal keybindings (input.conf) and
the multi-command feature (concatenating commands with ";"). It requires
a bunch of roundabout code to make it do the expected thing in
combination with async commands.
There is the question how commands should be handled that come in at a
higher rate than what can be handled by the core. Currently, it will
simply queue up input.conf commands as long as memory lasts. The client
API is limited by the size of the reply queue per client. For commands
which require a worker thread, the thread pool is limited to 30 threads,
and then will queue up work in memory. The number is completely
arbitrary.
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Fixes several issues playing back mpegts with video streams marked
as having "still images". For example, see this video which has
frames only every 6s: https://s3.amazonaws.com/tmm1/music-choice.ts
Changes include:
- start playback right away, without waiting for first video frame
- do not consider the sparse video stream in demuxer underrun detection
- do not require multiple video frames for the VO
- use audio as the master stream for demuxer metadata events
- use audio stream for playback time
Signed-off-by: Aman Gupta <aman@tmm1.net>
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Makes it easier to not break the build by confusing the ifdeffery.
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(Not sure if worth the trouble, but it does seem less awkward.)
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The purpose of the new API is to make it useable with other APIs than
OpenGL, especially D3D11 and vulkan. In theory it's now possible to
support other vo_gpu backends, as well as backends that don't use the
vo_gpu code at all.
This also aims to get rid of the dumb mpv_get_sub_api() function. The
life cycle of the new mpv_render_context is a bit different from
mpv_opengl_cb_context, and you explicitly create/destroy the new
context, instead of calling init/uninit on an object returned by
mpv_get_sub_api().
In other to make the render API generic, it's annoyingly EGL style, and
requires you to pass in API-specific objects to generic functions. This
is to avoid explicit objects like the internal ra API has, because that
sounds more complicated and annoying for an API that's supposed to never
change.
The opengl_cb API will continue to exist for a bit longer, but
internally there are already a few tradeoffs, like reduced
thread-safety.
Mostly untested. Seems to work fine with mpc-qt.
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Use the decoder wrapper that was introduced for video. This removes all
code duplication the old audio decoder wrapper had with the video code.
(The audio wrapper was copy pasted from the video one over a decade ago,
and has been kept in sync ever since by the power of copy&paste. Since
the original copy&paste was possibly done by someone who did not answer
to the LGPL relicensing, this should also remove all doubts about
whether any of this code is left, since we now completely remove any
code that could possibly have been based on it.)
There is some complication with spdif handling, and a minor behavior
change (it will restrict the list of codecs to spdif if spdif is to be
used), but there should not be any difference in practice.
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Move dec_video.c to filters/f_decoder_wrapper.c. It essentially becomes
a source filter. vd.h mostly disappears, because mp_filter takes care of
the dataflow, but its remains are in struct mp_decoder_fns.
One goal is to simplify dataflow by letting the filter framework handle
it (or more accurately, using its conventions). One result is that the
decode calls disappear from video.c, because we simply connect the
decoder wrapper and the filter chain with mp_pin_connect().
Another goal is to eventually remove the code duplication between the
audio and video paths for this. This commit prepares for this by trying
to make f_decoder_wrapper.c extensible, so it can be used for audio as
well later.
Decoder framedropping changes a bit. It doesn't seem to be worse than
before, and it's an obscure feature, so I'm content with its new state.
Some special code that was apparently meant to avoid dropping too many
frames in a row is removed, though.
I'm not sure how the source code tree should be organized. For one,
video/decode/vd_lavc.c is the only file in its directory, which is a bit
annoying.
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lavfi.c is not necessary anymore, because f_lavfi.c (which was actually
converted from it) can be used now.
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Get rid of the old vf.c code. Replace it with a generic filtering
framework, which can potentially handle more than just --vf. At least
reimplementing --af with this code is planned.
This changes some --vf se |
