Sometimes, --af=hrtf produces heavy artifacts or silence. It's possible
that this commit fixes these issues. My theory is that usually, the
uninitialized coefficients quickly converge to sane values as more audio
is filtered, which would explain why there are often artifacts on init,
with normal playback after that. It's also possible that sometimes, the
uninitialized values were NaN or inf, so that the artifacts (or silence)
would never go away.
Fix this by initializing the coefficients to 0. I'm not sure if this is
correct, but certainly better than before.
See issue #1104.
Let us set a different rate and delay.
Needed for the following commit where we set rate and delay reported by weston.
But only if the option native-keyrepeat is set.
Uses the new mechanism introduced in the previous commit.
Depending on the actual filter, this distributes CPU load more evenly
over time, although it probably doesn't matter.
Consider a filter which turns 1 frame into 2 frames (such as an
deinterlacer). Until now, we forced filters to produce all output frames
at once. This was done for simplicity.
Change the filter API such that a filter can produce frames
incrementally.
Rename video_decode_and_filter to video_filter, and add a new
video_decode_and_filter function. This function now calls the decoder.
This is done so that we can check filters a second time after decoding,
which avoids a useless playloop iteration.
(This and the previous commits are really just microoptimizations, which
simply reduce the number of times the playloop has to recheck
everything.)
Move the check to a function. Run the check a second time after
decoding/filtering. This second check is strictly speaking redundant
(which is why it wasn't done until now), but it avoids a useless
playloop iteration.
Move this code below the code that "shifts" the newly filtered frame.
This allows us to skip a useless playloop iteration later, because
obviously we need to filter a new frame after the previous frame has
been "shifted", and not before that.
Until now, you could override only level 3 with --osd-status-msg. Extend
this, add add --osd-msg1 to --osd-msg3 (one for each OSD level). OSD
level 0 always means disable OSD, so that isn't included.
--osd-msg3 corresponds to --osd-status-msg, but they're not exactly the
same. To allow more customization, --osd-msgN do not include the OSD
symbol. The symbol can be manually added with "${osd-sym-cc}". We keep
the "old" option for some short-term compatibility.
--osd-msg1 should be particularly useful; for example you could do:
--osd-msg1='${?pause==yes:${osd-sym-cc}}'
to display a "paused" symbol when paused, and nothing during normal
playback. (Although admittedly, the syntax is quite a bit of work.)
We don't allow this by default, because it would be silly if random
external data (like filenames or file tags) could accidentally trigger
them.
Add a property that magically disables this ASS tag escaping.
Note that malicious input could still disable ASS tag escaping by
itself. This would be annoying but harmless.
Pausing/unpausing while the audio device can't be reopened, and then
unpausing again when the device is finally reopened, can hang the
player for a while.
This happens because p->prepause_samples grows without bounds each
time the player is unpaused while the device is lost. On unpause,
ao_oss plays prepause_samples of silence to compensate for A/V timing
issues due to the partially lost buffer (we can't pause the device at
an arbitrary sample position, and the current period will be lost).
This in turn will make the player appear to be frozen if too much
audio is queued. (Normally, play() must never block, but here it
happens because more data is written than get_space() reports. A
better implementation would never let prepause_samples grow larger
than the period size.)
The unbounded growth happens because get_space() always returns that
the device can be written while the device is lost. So limit it to
200ms. (A better implementation would limit it to the period size.)
Also see #1080.
There's no reason to let the core wait until the frame is done
displaying. In practice, the core normally didn't need this additional
wakeup, and the VO was quick enough to fetch the new frame, before the
core even attempted to queue a new frame. But it wasn't entirely clean,
and the correct wakeup handling might matter in some cases.
With default settings, this allows you to hit the 100% mark (with
default --softvol-max in the middle) even if you've reached min or max
volume before. This is because 50 is not divisible by 3 (old default)
but by 2 (new default).
Not really sure why there still can be issues with higher --softvol-max
and --volstep=1, but this is where I stop caring.
If --write-filename-in-watch-later-config is used, and the filename
contains newline characters (as generally allowed on Unix), then the
newline will be written to the resume file literally, and the parts
after the newline character are interpreted as options.
This is possibly security relevant.
Change newline characters (and in fact any other special characters)
to '_'.
Reported as #1099 (this commit is a reimplementation of the proposed
pull request).
CC: @mpv-player/stable
Apparently making ESC exit fullscreen mode is the more popular
convention compared to ESC quitting the program.
It was also concluded that ESC should do nothing when the windows is
already in normal state.
See discussion in #973.
The filter output size can be 0. Due to how filtering works, this is
nothing unusual, but avresample_convert() will return 0. The same case
is already handling with "normal" resampling (this commit fixes the
reordering code).
Additionally, don't use an assert(). avresample_convert() failing is
unusual, but might also happen due to e.g. internal out of memory
conditions, so we shouldn't just crash on it.
Curiously observed with --ao=oss --audio-channels=5.1 when changing
speed.
Apparently NetBSD users want/need this (see issue #1080).
In order not to break playback, we need at least to emulate get_delay().
We do this approximately by using the system clock.
Also, always close the audio device on reset. Reopen it on play only. If
we can't reopen it, don't retry until after the next time reset or
resume is called, to avoid spam and unexpectedly "stealing" back the
audio device.
Also do something about framestepping causing audio desync.
The context struct had an audio_buf_info field, but there's no reason
why this would be needed. It's a tiny struct, and it isn't permanent
state. It's always returned by SNDCTL_DSP_GETOSPACE. Keeping this as
field is just confusing, so get rid of it.
The code for reopening the audio device was separate, and duplicated
some of the "real" open code. This was very badly done, and major
required parts of initialization were skipped. Fix this by removing
the code duplication. This consists mainly of moving the code for
opening the device to a separate function, and adding some changes
to handle format changes gracefully. (We can't change the audio
format on the fly, but we can at least not explode and play noise
when that happens.)
As a minor change, actually always use SNDCTL_DSP_RESET when closing
the audio device. We don't want to wait until the rest of the buffer
is played.
Also, don't use strerror() when printing the error message that
reopening failed, simply because reopen_device() takes care of this,
and also errno might be clobbered at this point.
I have no idea whether this is true, because there literally doesn't
seem to exist documentation for SNDCTL_DSP_RESET. But at least on
Linux' OSS emulation, it is true. Also, it would be quite insane if
it would be needed.
It seems on NetBSD SNDCTL_DSP_RESET exists, but using it for pausing
is not feasible. We still use it to discard the audio buffer when
closing the audio device.
Some window managers can prevent mapping of a window as a feature. i3
can put new windows on a certain workspace (with "assign"), so if mpv is
started on a different workspace, the window will never be mapped.
mpv currently waits until the window is mapped (blocking almost all of
the player), in order to avoid race conditions regarding the window
size. We don't want to remove this, but on the other hand we also don't
want to block the player forever in these situations.
So what we need is a way to know when the window manager is "done" with
processing the map request. Unfortunately, there doesn't seem to be a
standard way for this. So, instead we could do some arbitrary
communication with the WM, that may act as "barrier" after map request
and the "immediate" mapping of the window. If the window is not mapped
after this barrier, it means the window manager decided to delay the
mapping indefinitely. Use the _NET_REQUEST_FRAME_EXTENTS message as such
a barrier. WMs supporting this message must set the _NET_FRAME_EXTENTS
property on the mpv window, and we receive a PropertyNotify event. If
that happens, we always continue and cancel waiting for the MapNotify
event.
I don't know if this is sane or if there's a better mechanism. Also,
this works only for WMs which support this message, which are not many.
But at least it appears to work on i3. It may reintroduce flickering on
fullscreen with other WMs, though.
This is pretty imperfect, but it's just a demo.
The main purpose is clarifying how and when to get the video size.
In theory, retrieving the properties this way has a race condition:
after reading dwidth, the video could resize again. But the worst that
can happen are mismatching dwidth/dheight values, and the
MPV_EVENT_VIDEO_RECONFIG event would be immediately received again,
which would fix the mismatch. You could read the full video-out-params
property to absolutely avoid it, but it's not worth the trouble.
Merges pull request #1094, with some minor changes. mpv expects IEEE,
and IEEE allows divisions by 0 for floats, so these shouldn't actually
be a problem, but do it anyway for the sake of clang.
Signed-off-by: wm4 <wm4@nowhere>
Use libwaio to read from pipes (stdin or named pipes) on Windows. This
liberates us from nasty issues, such as pipes (as created by most
programs) not being possible to read in a non-blocking or event-driven
way. Although it would be possible to do that in a somewhat sane way
on Vista+, it's still not easy, and on XP it's especially hard. libwaio
handles these things for us.
Move pipe.c to pipe-unix.c, and remove Windows specific things. Also
adjust the input.c code to make this work cleanly.
