This simply writes the file name as a comment to the top of the watch later
config file.
It can be useful to the user for determining whether a watch later config file
can be manually removed (e.g. in case the corresponding media file has been
deleted) or not.
This accidentally rejected d==0. We can actually deal with sub-normals
fine, we just want to exclude nan and infinity (although infinity is
already accounted for, but anyway).
If a single person complains, I will readd it. But I don't expect that
this will happen.
The main reason for removing this is that it's some of the most unclean
code remaining, it's unmaintained, and I've never ever heard of someone
using it.
Until now, an error was reported only if the command couldn't be parsed.
Attempt to do more fine-grained reporting. This is not necessarily
perfect, but it's an improvement.
This would imply eglGetProcAddress() doesn't work correctly, but using
dlsym() does. For now get rid of it - it won't work in libmpv, and we'll
probably need a better workaround if it's still broken.
This code was in the initial wayland commit.
The functions glXGetProcAddressARB() and glXQueryExtensionsString() were
loaded using dlsym(). This could fail when compiling to libmpv, because
then dlopen(NULL, ...) will look in the main program's list of
libraries, and the libGL linked to libmpv is never considered. (Don't
know if this somehow could be worked around.) The result is that using
vo_opengl with libmpv can fail.
Avoid this by not using dlsym(). glXGetProcAddressARB() was already used
directly in the same file, and that never caused any problems. (Still
add it to the configure test.) glXQueryExtensionsString() is documented
as added in GLX 1.1 - that's ancient.
So the device buffer can be refilled quickly. Fixes dropouts in certain
cases: if all data is moved from the soft buffer to the audio device
buffer, the waiting code thinks it has to enter the mode in which it
waits for new data from the decoder. This doesn't work, because the
get_space() logic tries to keep the total buffer size down. get_space()
will return 0 (or a very low value) because the device buffer is full,
and the decoder can't refill the soft buffer. But this means if the AO
buffer runs out, the device buffer can't be refilled from the soft
buffer. I guess this mess happened because the code is trying to deal
with both AOs with proper event handling, and AOs with arbitrary
behavior.
Unfortunately this increases latency, as the total buffered audio
becomes larger. There are other ways to fix this again, but not today.
Fixes#818.
Apparently this can happen. So actually only return from waiting if ALSA
excplicitly signals that new output is available, or if we are woken up
externally.
This call was used limited the buffer size if installed RAM was below 16
MB. This stopped being useful a decade ago. The check could also
overflow on 32 bit systems. Just get rid of it.
This did not flush remaining audio in the buffer correctly (in case an
AO has an internal block size). So we have to make the audio feed thread
to write the remaining audio, and wait until it's done.
Checking the avoid_ao_wait variable should be enough to be sure that all
data that can be written was written to the AO driver.
This code handles buggy AOs (even if all AOs are bug-free, it's good for
robustness). Move handling of it to the AO feed thread. Now this check
doesn't require magic numbers and does exactly what's it supposed to do.
Until now, we've always calculated a timeout based on a heuristic when
to refill the audio buffers. Allow AOs to do it completely event-based
by providing wait and wakeup callbacks.
This also shuffles around the heuristic used for other AOs, and there is
a minor possibility that behavior slightly changes in real-world cases.
But in general it should be much more robust now.
ao_pulse.c now makes use of event-based waiting. It already did before,
but the code for time-based waiting was also involved. This commit also
removes one awkward artifact of the PulseAudio API out of the generic
code: the callback asking for more data can be reentrant, and thus
requires a separate lock for waiting (or a recursive mutex).
uint_least32_t could be larger than uint32_t, so the return values of
mp_ring_get_wpos/rpos must be adjusted. Actually just use unsigned long
as type instead, because that is less awkward than uint_least32_t.
This was originally added because we thought this would make a good
portable audio API, which would give us good behavior on Windows, Linux,
and OSX. But this hope was disappointed: it's not reliable enough (nice
deadlocks on Linux when seeking, i.e. resetting the audio device),
doesn't have enough features (no channel maps, no digital passthrough),
and in general just is not very good.
There were subtle and minor race conditions in the pull.c code, and AOs
using it (jack, portaudio, sdl, wasapi). Attempt to remove these.
There was at least a race condition in the ao_reset() implementation:
mp_ring_reset() was called concurrently to the audio callback. While the
ringbuffer uses atomics to allow concurrent access, the reset function
wasn't concurrency-safe (and can't easily be made to).
Fix this by stopping the audio callback before doing a reset. After
that, we can do anything without needing synchronization. The callback
is resumed when resuming playback at a later point.
Don't call driver->pause, and make driver->resume and driver->reset
start/stop the audio callback. In the initial state, the audio callback
must be disabled.
JackAudio of course is different. Maybe there is no way to suspend the
audio callback without "disconnecting" it (what jack_deactivate() would
do), so I'm not trying my luck, and implemented a really bad hack doing
active waiting until we get the audio callback into a state where it
won't interfere. Once the callback goes from AO_STATE_WAIT to NONE, we
can be sure that the callback doesn't access the ringbuffer or anything
else anymore. Since both sched_yield() and pthread_yield() apparently
are not always available, use mp_sleep_us(1) to avoid burning CPU during
active waiting.
The ao_jack.c change also removes a race condition: apparently we didn't
initialize _all_ ao fields before starting the audio callback.
In ao_wasapi.c, I'm not sure whether reset really waits for the audio
callback to return. Kovensky says it's not guaranteed, so disable the
reset callback - for now the behavior of ao_wasapi.c is like with
ao_jack.c, and active waiting is used to deal with the audio callback.
We don't need to combine __sync_add_and_fetch with a memory barrier,
since these intrinsics are documented as using a full barrier already.
Use __sync_fetch_and_add instead of __sync_add_and_fetch; this gives
atomic_fetch_add() the correct return value (although we don't use it).
Use __sync_fetch_and_add to emulate atomic_load(). This should enforce
the full barrier semantics better. (This trick is stolen from the
FreeBSD based stdatomic.h emulation.)
In most places where af_fmt2bits is called to get the bits/sample, the
result is immediately converted to bytes/sample. Avoid this by getting
bytes/sample directly by introducing af_fmt2bps.
The i_bps members of the sh_audio and dev_video structs are mostly used
for displaying the average audio and video bitrates. Keeping them in
bits-per-second avoids truncating them to bytes-per-second and changing
them back lateron.
This is incomplete; the video chain will still hold some vaapi objects
after destroying the decoder and thus the vaapi context. This is very
bad. Fixing it would require something like refcounting the vaapi
context, but I don't really want to.
mpv supports two hardware decoding APIs on Linux: vdpau and vaapi. Each
of these has emulation wrappers. The wrappers are usually slower and
have fewer features than their native opposites. In particular the libva
vdpau driver is practically unmaintained.
Check the vendor string and print a warning if emulation is detected.
Checking vendor strings is a very stupid thing to do, but I find the
thought of people using an emulated API for no reason worse.
Also, make --hwdec=auto never use an API that is detected as emulated.
This doesn't work quite right yet, because once one API is loaded,
vo_opengl doesn't unload it, so no hardware decoding will be used if the
first probed API (usually vdpau) is rejected. But good enough.
So, basically this worked only with streams that were not local files,
because stream_dvd.c "intercepts" local files to check whether they
point to DVD images. This means if a stream is not writeable, we have to
try the next stream implementation.
Unbreaks 2-pass encoding.
Sometimes, Matroska files store monotonic PTS for h264 tracks with
b-frames, which means the decoder actually returns non-monotonic PTS.
Handle this with an evil trick: if DTS is missing, set it to the PTS.
Then the existing logic, which deals with falling back to DTS if PTS is
broken. Actually, this trick is not so evil at all, because usually, PTS
has no errors, and DTS is either always set, or always unset. So this
_should_ provoke no regressions (famous last words).
libavformat actually does something similar: it derives DTS from PTS in
ways unknown to me. The result is very broken, but it causes the DTS
fallback to become active, and thus happens to work.
Also, prevent the heuristic from being active if PTS is merely monotonic
instead of strictly-monotonic. Non-unique PTS is broken, but we can't
fallback to DTS anyway in these cases.
The specific mkv file that is fixed with this commit had the following
fields set:
Muxing application: libebml v1.3.0 + libmatroska v1.4.1
Writing application: mkvmerge v6.7.0 ('Back to the Ground') [...]
But I know that this should also fix playback of mencoder produced mkv
files.
