| Commit message (Collapse) | Author | Age | Files | Lines |
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On a audio/video desync by more than 0.5 seconds, display-sync mode was
disabled, and not enabled again (until playback restart, e.g. a seek).
The idea was that it this only happens when this playback mode is broken
and can't perform well anyway (A/V desync is a clear indication that
something is very wrong). Instead of behaving like a god damn POS, it
should revert to the more robust audio-sync mode.
Unfortunately, this could happen sporadically due to temporary system
performance problems, such as toggling fullscreen. Users didn't like
this, and asked for a function to disable it, or to recover in some
other way.
This mechanism is questionable anyway. If an ignorant user enables
display-sync, and encounters problems with it (without being able to
determine that display-sync is messing up), the player will still behave
like a POS on every playback, and even after every seek. It might
actually be helpful to fail more consistently. Also, I've found that
it's sill relatively reliable anyway even without this mechanism.
So just remove the fallback.
Fixes: #7048
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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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Unless --video-latency-hacks, always decode 2 frames on playback
restart. This in turn will always compute the correct frame duration
(even for the first frame), which in turn happens to fix that playback
with an image at the beginning breaks display.
If a still image precedes video, and the size/format of the frame is
different from that of the video following it, the incorrect frame
duration caused vo_reconfig2() to be called early, causing the window to
resize, and the renderer to clear the image to black. Specifically, it
hit the default value of 1 second duration (for still images), so the
image was displayed for 1 second, and changed to black until the next
proper video frame was displayed.
Normally this does not happen. Even if a video file displays still
images, it normally repeats the still image at the video's FPS (which is
sane). But you can construct such files, or use EDL to construct
something similarly behaving.
This change may increase seek latency a bit in audio video-sync mode
(the default). It needs to wait until 2 frames are decoded, before it
bothers to display the first frame. This is done even when seeking. In
theory it might be good to introduce a "seek preview" mode, which shows
the target image without all the preparations needed for starting
playback. (For example, it could not decode audio.) But since I'm using
video-sync=display-resample, which already needed to always decode 2
frames, I don't think this is a terribly high priority, nor do I
consider the slightly slower seeking a regression.
Fixes: #6765
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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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We need to transform the timestamp returned by get_play_end_pts().
I considered making it return the transformed timestamp directly. There
are 4 callers; 2 need a transformed timestamps, 2 don't. So I guess it
doesn't matter.
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E.g. "mpv null:// --demuxer=rawvideo" will "hang" by waiting for video
EOF forever. It's not signalled correctly because of the last-frame
corner case, which attempts to wait until the current frame is finally
displayed (which is signalled by whether a new frame can be queued, see
commit 1a339fa09d for some details). If no frame was ever queued, the VO
is not configured, and vo_is_ready_for_frame() never returns true.
Fix this by using vo_has_frame(), which seems to be exactly the correct
thing we need.
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If the container FPS is correct, this can help getting ideal mix factors
for vo_gpu interpolation mode. Otherwise, it doesn't matter.
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Basically reimplement the async behavior on top of the async command
code. With this, all screenshot commands are async, and the "async"
prefix basically does nothing. The prefix now behaves exactly like with
other commands that use spawn_thread.
This also means using the prefix in the preset input.conf is pointless
(without effect) and misleading, so remove that.
The each_frame mode was actually particularly painful in making this
change, since the player wants to block for it when writing a
screenshot, and generally doesn't fit into the new infrastructure. It
was still relatively easy to reimplement by copying the original command
and then repeating it on each frame. The waiting is reentrant now, so
move the call in video.c to a "safer" spot.
One way to observe how the new semantics interact with everything is
using the mpv repl script and sending a screenshot command through it.
Without async flag, the script will freeze while writing the screenshot
(while playback continues), while with async flag it continues.
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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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Until recently, ao_lavc and vo_lavc started encoding whenever the core
happened to send them data. Since audio and video are not initialized at
the same time, and the muxer was not necessarily opened when the first
encoder started to produce data, the resulting packets were put into a
queue. As soon as the muxer was opened, the queue was flushed.
Change this to make the core wait with sending data until all encoders
are initialized. This has the advantage that we don't need to queue up
the packets.
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The video timing code could just decide that EOF was reached before it
was displayed. This is not really a problem for normal playback (if you
use something like --keep-open it'd show the last frame anyway,
otherwise it'd at best flash it on screen before destroying the window).
But in encode mode, it really matters, and makes the difference between
having one frame more or less in the output file.
Fix this by waiting for the VO before starting the real EOF.
vo_is_ready_for_frame() is normally used to determine when the VO frame
queue has enough space to send a new frame. Since the VO frame queue is
currently at most 1 frame, it being signaled means the remaining frame
was consumed and thus sent to the VO driver. If it returns false, it
will wake up the playloop as soon as the state changes.
I also considered using vo_still_displaying(), but it's not reliable,
because it checks the realtime of the frame end display time.
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This code makes the player wait using real time, which makes sense for
normal playback, but not encode mode.
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The main change is that we wait with opening the muxer ("writing
headers") until we have data from all streams. This fixes race
conditions at init due to broken assumptions in the old code.
This also changes a lot of other stuff. I found and fixed a few API
violations (often things for which better mechanisms were invented, and
the old ones are not valid anymore). I try to get away from the public
mutex and shared fields in encode_lavc_context. For now it's still
needed for some timestamp-related fields, but most are gone. It also
removes some bad code duplication between audio and video paths.
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1. I want to get away from mp_image_params (maybe).
2. For encoding mode, it's convenient to get the nominal_fps, which is
a mp_image field, and not in mp_image_params.
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There is some sort-of awkwardness here, because option access needs to
happen in a synchronized manner, and the framedrop flag is not in the VO
option struct. Remove the mp_read_option_raw() call and the awkward
change notification via VO_EVENT_WIN_STATE from command.c, and pass it
through as new vo_frame flag.
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The playback start logic explicitly waits until the first frame has been
displayed. Usually this will introduce a wait of 1 vsync. For normal
playback this doesn't matter, but with respect to low latency needs,
this only leads to additional data getting queued up in the demuxer or
network buffers.
Another thing is that the timing logic decodes 1 frame ahead (= 1 frame
extra latency) to determine the exact duration of a frame.
To be fair, there doesn't really seem to be a hard reason why this is
needed. With the current code, enabling the option does lead to A/V
desync sometimes (if the demuxer FPS is too inaccurate), and also frame
drops at playback start in some situations. But this all seems to be
avoidable, if the timing logic were to be rewritten completely, which
should probably happen in the future. Thus the new option comes with the
warning that it can be removed any time. This is also why the option has
"hack" in the name.
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The extra frame is used to compute the exact frame duration. But frame
drop is disabvled with --untimed.
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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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This fixes playback stalls on some mediacodec hardware decoders,
which expect that frame buffers will be rendered and returned back
to the decoder as soon as possible.
Specifically, the issue was observed on an NVidia SHIELD Android TV,
only when playing an H264 sample which switched between interlaced
and non-interlaced frames. On an interlacing change, the decoder
expects all outstanding frames would be returned to it before it
would emit any new frames. Since a single extra frame always remained
buffered by mpv, playback would stall. After this commit, no extra
frames are buffered by mpv when using vo_mediacodec_embed.
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To make this less of a mess, remove one of the redundant container_fps
fields.
Part of #5470.
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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 semantics (including runtime behavior and the
"vf" command). The most important ones are listed in interface-changes.
vf_convert.c is renamed to f_swscale.c. It is now an internal filter
that can not be inserted by the user manually.
f_lavfi.c is a refactor of player/lavfi.c. The latter will be removed
once --lavfi-complex is reimplemented on top of f_lavfi.c. (which is
conceptually easy, but a big mess due to the data flow changes).
The existing filters are all changed heavily. The data flow of the new
filter framework is different. Especially EOF handling changes - EOF is
now a "frame" rather than a state, and must be passed through exactly
once.
Another major thing is that all filters must support dynamic format
changes. The filter reconfig() function goes away. (This sounds complex,
but since all filters need to handle EOF draining anyway, they can use
the same code, and it removes the mess with reconfig() having to predict
the output format, which completely breaks with libavfilter anyway.)
In addition, there is no automatic format negotiation or conversion.
libavfilter's primitive and insufficient API simply doesn't allow us to
do this in a reasonable way. Instead, filters can use f_autoconvert as
sub-filter, and tell it which formats they support. This filter will in
turn add actual conversion filters, such as f_swscale, to perform
necessary format changes.
vf_vapoursynth.c uses the same basic principle of operation as before,
but with worryingly different details in data flow. Still appears to
work.
The hardware deint filters (vf_vavpp.c, vf_d3d11vpp.c, vf_vdpaupp.c) are
heavily changed. Fortunately, they all used refqueue.c, which is for
sharing the data flow logic (especially for managing future/past
surfaces and such). It turns out it can be used to factor out most of
the data flow. Some of these filters accepted software input. Instead of
having ad-hoc upload code in each filter, surface upload is now
delegated to f_autoconvert, which can use f_hwupload to perform this.
Exporting VO capabilities is still a big mess (mp_stream_info stuff).
The D3D11 code drops the redundant image formats, and all code uses the
hw_subfmt (sw_format in FFmpeg) instead. Although that too seems to be a
big mess for now.
f_async_queue is unused.
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I've decided that MP_TRACE means “noisy spam per frame”, whereas
MP_DBG just means “more verbose debugging messages than MSGL_V”.
Basically, MSGL_DBG shouldn't create spam per frame like it currently
does, and MSGL_V should make sense to the end-user and provide mostly
additional informational output.
MP_DBG is basically what I want to make the new default for --log-file,
so the cut-off point for MP_DBG is if we probably want to know if for
debugging purposes but the user most likely doesn't care about on the
terminal.
Also, the debug callbacks for libass and ffmpeg got bumped in their
verbosity levels slightly, because being external components they're a
bit less relevant to mpv debugging, and a bit too over-eager in what
they consider to be relevant information.
I exclusively used the "try it on my machine and remove messages from
MSGL_* until it does what I want it to" approach of refactoring, so
YMMV.
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update_subtitles() makes sure all subtitle packets at/before the given
PTS have been read and processed. Normally, this function is only called
before sending a frame to the VO. This is too late for vf_sub, which
expects the subtitles to be updated before feeding a frame to the
filters.
Apparently this was specifically a problem for the first frame.
Subsequent frames might have been ok due to general prefetching.
(This will fail anyway, should a filter dare to add an offset to the
timestamps of the filered frames before they pass to vf_sub.)
Fixes #5194.
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The internal stereo3d filter was removed due to being GPL only, and due
to being a mess that somehow used libavfilter's filter. Without this
filter, it's hard to remove our internal stereo3d image attribute, so
even using libavfilter's stereo3d filter would not work too well (unless
someone fixes it and makes it able to use AVFrame metadata, which we
then could mirror in mp_image).
This was never well thought-through anyway, so just drop it. I think
some "downsampling" support would still make sense, maybe that can be
readded later.
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Now using libavfilter filters directly.
The rotation case is a bit lazy, because it uses the slow vf_rotate
filter in all cases, instead of using special filters for 90° step
rotations.
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Regression introduced by direct rendering code additions. Found by same
static analyzer.
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This code could not be relicensed. The intention was to write new filter
code (which could handle both audio and video), but that's a bit of
work. Write some code that can do audio conversion (resampling,
downmixing, etc.) without the old audio filter chain code in order to
speed up the LGPL relicensing.
If you build with --disable-libaf, nothing in audio/filter/* is compiled
in. It breaks a few features, such as --volume, --af, pitch correction
on speed changes, replaygain.
Most likely this adds some bugs, even if --disable-libaf is not used.
(How the fuck does EOF notification work again anyway?)
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This removes all GPL only code from it, and that's the whole purpose.
Also happens to be much simpler.
The "deinterlace" option still sort of exists, but only as runtime
changeable option. The main change in behavior is that the property will
not report back the actual deint state. Or in other words, if inserting
or initializing the filter fails, the deinterlace property will still
return "yes". This is in line with most recent behavior changes to
properties and options.
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I really wouldn't care much about this, but some parts of the core code
are under HAVE_GPL, so there's some need to get rid of it. Simply turn
the video equalizer from its current fine-grained handling with vf/vo
fallbacks into global options. This makes updating them much simpler.
This removes any possibility of applying video equalizers in filters,
which affects vf_scale, and the previously removed vf_eq. Not a big
loss, since the preferred VOs have this builtin.
Remove video equalizer handling from vo_direct3d, vo_sdl, vo_vaapi, and
vo_xv. I'm not going to waste my time on these legacy VOs.
vo.eq_opts_cache exists _only_ to send a VOCTRL_SET_EQUALIZER, which
exists _only_ to trigger a redraw. This seems silly, but for now I feel
like this is less of a pain. The rest of the equalizer using code is
self-updating.
See commit 96b906a51d5 for how some video equalizer code was GPL only.
Some command line option names and ranges can probably be traced back to
a GPL only committer, but we don't consider these copyrightable.
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This is pretty pointless, but I believe it allows us to claim that the
new code is not affected by the copyright of the old code. This is
needed, because the original mp_audio struct was written by someone who
has disagreed with LGPL relicensing (it was called af_data at the time,
and was defined in af.h).
The "GPL'ed" struct contents that surive are pretty trivial: just the
data pointer, and some metadata like the format, samplerate, etc. - but
at least in this case, any new code would be extremely similar anyway,
and I'm not really sure whether it's OK to claim different copyright. So
what we do is we just use AVFrame (which of course is LGPL with 100%
certainty), and add some accessors around it to adapt it to mpv
conventions.
Also, this gets rid of some annoying conventions of mp_audio, like the
struct fields that require using an accessor to write to them anyway.
For the most part, this change is only dumb replacements of mp_audio
related functions and fields. One minor actual change is that you can't
allocate the new type on the stack anymore.
Some code still uses mp_audio. All audio filter code will be deleted, so
it makes no sense to convert this code. (Audio filters which are LGPL
and which we keep will have to be ported to a new filter infrastructure
anyway.) player/audio.c uses it because it interacts with the old filter
code. push.c has some complex use of mp_audio and mp_audio_buffer, but
this and pull.c will most likely be rewritten to do something else.
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Refresh seeks are |
