Adds always-on mode by internally utilizing hidetimeout as negative and
forbidding the user to set negative values.
This removes script-message to enable/disable the osc, and instead introduces a
combined 'visibility' control with the values never/auto/always.
It's available via script_opts and script_message as 'osc-visibility'.
As message, it also supports a 'cycle' value.
The del key is bound to cycling the visibility modes.
There were few issues:
- When it's disabled and then enabled, it was displaying the osc briefly and
then autohide right away. Don't do that.
- When it's enabled and then disabled, it was not removing the osc from screen
if called while the osc is visible (because tick() is responsible for the hide
but it doesn't render() the empty osc when the osc is disabled).
- Due to delayed/async unbinding of mouse events it was possible to show_osc()
after it got disabled e.g. from mouse_move. Prevent this.
This eliminates some intermittent pops heard in a HRT MicroStreamer DAC
uncorrelated with user interaction. As a bonus, this resolves#1773 which I can
o longer reproduce as of this commit. Leave the 50ms buffer for shared mode
since that seems to be working quite well.
This is also the way exclusive mode is done in the MSDN example code:
https://msdn.microsoft.com/en-us/library/windows/desktop/dd370844%28v=vs.85%29.aspx
This was originally increased in c545c40 to mitigate glitches that subsequent
refactorings have eliminated.
A COM message loop is apparently totally inappropriate for a low latency
thread. It leads to audio glitches because the thread doesn't wake up fast
enough when it should. It also causes mysterious correlations between the vo
and ao thread (i.e., toggling fullscreen delays audio feed events). Instead use
an mp_dispatch_queue to set/get volume/mute/session display name from the audio
thread. This has the added benefit of obviating the need to marshal the
associated interfaces from the audio thread.
Don't wait for WASAPI to send another feed event if we detect an underfull
buffer. It seems that WASAPI doesn't always send extra feed events if
something causes rendering to fall behind. This causes every subsequent playback
buffer to under-run until playback is reset. The fix is simply to do a one-shot
double feed when this happens, which allows rendering to catch up with playback.
This was observed to happen when using MsgWaitForMultipleObjects to wait for the
feed event and toggling fullscreen with vo=opengl:backend=win. This commit
improves the behaviour in that specific case and more generally makes exclusive
mode significantly more robust.
This commit also moves the logic to avoid *over*filling the exclusive mode
buffer into thread_feed right next to the above described underfil logic.
OpenSL ES is used on Android. At the moment only stereo output is
supported. Two options are supported: 'frames-per-buffer' and
'sample-rate'. To get better latency the user of libmpv should pass
values obtained from AudioManager.getProperty(PROPERTY_OUTPUT_FRAMES_PER_BUFFER)
and AudioManager.getProperty(PROPERTY_OUTPUT_SAMPLE_RATE).
_Of course_ the previous commit broke --force-window behavior (like it
does every single time I touch it).
vo_has_frame() gets cleared after a seek, so e.g. stopping playback of a
file and going to the next by keeping the seek key down will enter a
short moment without video at the end of the first file, which will set
the stalled_video variable to true. Prevent it by using the indication
whether the window was properly created (which is probably exactly what
we want here).
This function is also responsible for destroying the window when needed,
and obviously we should never do that while video is active. (This is
the actual bug, although the other change in this commit already hides
the common breakage it caused.)
If a stream is marked as EOF (due to no packets found in reach), then we
need to wakeup the decoder. This is important especially if no packets
are found at the start of the file, so the A/V sync logic actually
starts playback, instead of waiting for packets that will never come.
(It would randomly start playback when running the playback loop due to
arbitrary external events like user input.)
Some oddity that is not needed anymore. The only thing which still
referenced them was avoiding loading external files more than once,
which is now prevented by checking the list of tracks instead.
Why was this done so stupidly, with so many complicated special cases,
before? Declare it once so the shader bits don't have to figure out where
and when to do so themselves.
Commit 943f76e6, which already tried this, was very stupid: it didn't
actually override the samplerate for Opus, but overrode it for all
codecs other than Opus. And even then, it failed to use the overridden
samplerate. (Sigh...)
Fixes relative seeks. Without this, a seek back could skip so much data
that the seek would effectively jump forward. (Or insert silence for
files with video.)
There's the question whether the frontend should do this instead (by
using information from the decoders), but for now this seems more
proper.
demux_mkv.c does this already, sort of.
libavformat doesn't for seeks in .ogg (aka .opus), but might be doing it
for mkv. Seems to be a mess as well.
Fixes correctness_trimming_nobeeps.opus. One nasty thing is that this
mechanism interferes with the container-signalled mechanism with
AV_FRAME_DATA_SKIP_SAMPLES. So apply it only if that is apparently not
present. It's a mess, and it's still broken in FFmpeg CLI, so I'm sure
this will get fucked up later again.
I'm not quite sure what the FFmpeg AV_FRAME_DATA_SKIP_SAMPLES API
demands here. The code so far assumed that skipping can be more than a
frame, but not trimming. Extend it to trimming too.
This is actually already done by dec_audio.c. But if
AV_FRAME_DATA_SKIP_SAMPLES is applied, this happens too late here. The
problem is that this will slice off samples, and make it impossible for
later code to reconstruct the timestamp properly.
Missing timestamps can still happen with some demuxers, e.g. demux_mkv.c
with Opus tracks. (Although libavformat interpolates these itself.)
I think the conclusion is that AV_PKT_DATA_SKIP_SAMPLES is misdesigned
(at least for some formats), and an alternative mechanism using
durations would be better. (Combining it with a proper timebase would
keep sample-accuracy.)
This happens only if the new segment wasn't read yet.
This is not quite proper and a problem with dec_sub.c internals.
Ideally, it'd wait with rendering until a new enough segment has been
read. Normally, the new segment is available immediately, so the end
will be automatically clipped by switching to the right segment in the
exact moment it's supposed to become effective.
Usually shouldn't cause any problems, though.
Doing --hwdec=auto ends up picking dxva2, creating a decoder, and then
sending D3D frames down the video chain, which immediately fails and
falls back to software.
Consider dxva2 only if the VO provides a context. If this fails,
autoprobing will proceed to try dxva2-copy as usual.
Fixes#2844.
igv
Avi Halachmi (:avih)
wm4
Kevin Mitchell
Ilya Zhuravlev
igv
Niklas Haas
Gusar321
Emmanuel Gil Peyrot