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.
This does what af_volume used to do. Since we couldn't relicense it,
just rewrite it. Since we don't have a new filter mechanism yet, and the
libavfilter is too inconvenient, do applying the volume gain in ao.c
directly. This is done before handling the audio data to the driver.
Since push.c runs a separate thread, and pull.c is called asynchronously
from the audio driver's thread, the volume value needs to be
synchronized. There's no existing central mutex, so do some shit with
atomics. Since there's no atomic_float type predefined (which is at
least needed when using the legacy wrapper), do some nonsense about
reinterpret casting the float value to an int for the purpose of atomic
access. Not sure if using memcpy() is undefined behavior, but for now I
don't care.
The advantage of not using a filter is lower complexity (no filter auto
insertion), and lower latency (gain processing is done after our
internal audio buffer of at least 200ms).
Disavdantages include inability to use native volume control _before_
other filters with custom filter chains, and the need to add new
processing for each new sample type.
Since this doesn't reuse any of the old GPL code, nor does indirectly
rely on it, volume and replaygain handling now works in LGPL mode.
How to process the gain is inspired by libavfilter's af_volume (LGPL).
In particular, we use exactly the same rounding, and we quantize
processing for integer sample types by 256 steps. Some of libavfilter's
copyright may or may not apply, but I think not, and it's the same
license anyway.
Just reimplement it in some way, as mp_audio is GPL-only.
Actually I wanted to get rid of audio_buffer.c completely (and instead
have a list of mp_aframes), but to do so would require rewriting some
more player core audio code. So to get this LGPL relicensing over
quickly, just do some extra work.
While this is perfectly OK on Unix, it causes annoying valgrind
warnings, and might be otherwise confusing to others.
On Windows, the runtime can actually abort the process if this is
called.
push.c part taken from a patch by Pedro Pombeiro.
Before this change, AOs could have internal alignment, and play() would
not consume the trailing data if the size passed to it is not aligned.
Change this to require AOs to report their alignment (via period_size),
and make sure to always send aligned data.
The buffer reported by get_space() now always has to be correct and
reliable. If play() does not consume all data provided (which is bounded
by get_space()), an error is printed.
This is preparation for potential further AO changes.
I casually checked alsa/lavc/null/pcm, the other AOs might or might not
work.
Until now, this was only implemented for ao_alsa and AOs not using
push.c. ao_alsa.c relied on enabling funny underrun semantics for
avoiding resets on lower levels, while other AOs using push.c didn't do
anything.
Change this and at least make push.c copy silent data to the AO. This
still isn't perfect as keeping track of how much silence was played when
seems complex, so we don't do it. The consequence is that frame-stepping
will essentially randomize the A/V offset (it'll recover immediately
when unpausing, but still ugly). Also, in order to empty the currently
buffered audio on seeks etc., we still call ao_driver->reset and so on,
so the AO driver will still need to handle this specially.
The intent is to make behavior with ALSA less weird (for one we can
remove the code in ao_alsa.c that tries to trigger an initial
underflow). Also might help with #3754.
The player tries to avoid splitting frames with spdif (sample alignment
stuff). This can in certain corner cases with certain drivers lead to
the situation that ao_get_space() returns a number higher than 0 and
lower than the audio frame size. The playloop will round this down to 0
bytes and do nothing, leading to a missed wakeup. This can lead to
underruns or playback completely getting stuck.
It can be reproduced by playing AC3 passthrough with no video and:
--ao=null --ao-null-buffer=0.256 --ao-null-outburst=6100
This commit attempts to fix it by allowing the playloop to write some
additional data (to get a complete frame), that will be buffered within
the AO ringbuffer even if the audio device doesn't want it.
Currently, calling mp_input_wakeup() will wake up the core thread (also
called the playloop). This seems odd, but currently the core indeed
calls mp_input_wait() when it has nothing more to do. It's done this way
because MPlayer used input_ctx as central "mainloop".
This is probably going to change. Remove direct calls to this function,
and replace it with mp_wakeup_core() calls. ao and vo are changed to use
opaque callbacks and not use input_ctx for this purpose. Other code
already uses opaque callbacks, or has legitimate reasons to use
input_ctx directly (such as sending actual user input).
This helps with shitty APIs and even shittier drivers (I'm looking at
you, ALSA). Sometimes they won't send proper wakeups. This can be fine
during playback, when for example playing video, because mpv still will
wakeup the AO outside of its own wakeup mechanisms when sending new data
to it. But when draining, it entirely relies on the driver's wakeup
mechanism. So when the driver wakeup mechanism didn't work, it could
hard freeze while waiting for the audio thread to play the rest of the
data.
Avoid this by waiting for an upper bound. We set this upper bound at the
total mpv audio buffer size plus 1 second. We don't use the get_delay
value, because the audio API could return crap for it, and we're being
paranoid here. I couldn't confirm whether this works correctly, because
my driver issue fixed itself.
(In the case that happened to me, the driver somehow stopped getting
interrupts. aplay froze instead of playing audio, and playing audio-only
files resulted in a chop party. Video worked, for reasons mentioned
above, but drainign froze hard. The driver problem was solved when
closing all audio output streams in the system. Might have been a dmix
related problem too.)
When we're draining, don't wakeup the core on every buffer fill, since
unlike during normal playback, we won't actually get more data. The
wakeup here conceptually works like wakeups with condition variables, so
redundant wakeups do not hurt, so this is just a minor change and
nothing of consequence.
(Final EOF also requires waking up the core, but there is separate code
to send this notification.)
Also dump the p->still_playing field in trace logging.
For clang, it's enough to just put (void) around usages we are
intentionally ignoring the result of.
Since GCC does not seem to want to respect this decision, we are forced
to disable the warning globally.
This covers source files which were added in mplayer2 and mpv times
only, and where all code is covered by LGPL relicensing agreements.
There are probably more files to which this applies, but I'm being
conservative here.
A file named ao_sdl.c exists in MPlayer too, but the mpv one is a
complete rewrite, and was added some time after the original ao_sdl.c
was removed. The same applies to vo_sdl.c, for which the SDL2 API is
radically different in addition (MPlayer supports SDL 1.2 only).
common.c contains only code written by me. But common.h is a strange
case: although it originally was named mp_common.h and exists in MPlayer
too, by now it contains only definitions written by uau and me. The
exceptions are the CONTROL_ defines - thus not changing the license of
common.h yet.
codec_tags.c contained once large tables generated from MPlayer's
codecs.conf, but all of these tables were removed.
From demux_playlist.c I'm removing a code fragment from someone who was
not asked; this probably could be done later (see commit 15dccc37).
misc.c is a bit complicated to reason about (it was split off mplayer.c
and thus contains random functions out of this file), but actually all
functions have been added post-MPlayer. Except get_relative_time(),
which was written by uau, but looks similar to 3 different versions of
something similar in each of the Unix/win32/OSX timer source files. I'm
not sure what that means in regards to copyright, so I've just moved it
into another still-GPL source file for now.
screenshot.c once had some minor parts of MPlayer's vf_screenshot.c, but
they're all gone.
Replace all the check macros with function calls. Give them all the
same case and naming schema.
Drop af_fmt2bits(). Only af_fmt2bps() survives as af_fmt_to_bytes().
Introduce af_fmt_is_pcm(), and use it in situations that used
!AF_FORMAT_IS_SPECIAL. Nobody really knew what a "special" format
was. It simply meant "not PCM".
There is not much of a reason to have these wrappers around. Use POSIX
standard functions directly, and use a separate utility function to take
care of the timespec calculations. (Course POSIX for using this weird
format for time values.)
If calling ao->driver->wait() fails, we need to fallback to timeout-
based waiting. But it could be that at this point, the mutex was already
released (and then re-acquired). So we need to recheck the condition in
order to avoid missed wakeups.
This probably wasn't an actually occurring problem, but still could
cause a small race-condition window if the dynamic fallback is actually
used.
Especially with other components (libavcodec, OSX stuff), the thread
list can get quite populated. Setting the thread name helps when
debugging.
Since this is not portable, we check the OS variants in waf configure.
old-configure just gets a special-case for glibc, since doing a full
check here would probably be a waste of effort.
Don't wait after the audio thread has pushed the remaining audio to the
AO. Avoids hard hangs if the heuristic fails completely (could still
happen if get_delay returns absurd values).
CC: @mpv-player/stable
Since the internal AO driver API has no proper way to determine EOF, we
need to guess by querying get_delay. But some AOs (e.g. ao_pulse with
no-latency-hacks set) may never reach 0, maybe because they naively add
the latency to the buffer level. In this case our heuristic can break.
Fix by always using the delay to estimate the EOF time. It's not even
that important - it's mostly used to avoid blocking draining. So this
should be ok.
CC: @mpv-player/stable (maybe)
It was more complicated than it had to be: the audio thread already
determines whether audio has ended, so we can use that. Remove the
separate logic for draining.
Improve the logic how the audio thread decides how to wait until the AO
is ready for new data. The previous commit makes some of this easier,
although it turned out that it wasn't required, and we still can handle
AOs with bad get_space implementation (although the new code prints an
error message, and it might fail in obscure situations).
The new code is pretty similar to the old one, and the main thing that
changes is that complicated conditions are tweaked. AO waiting is now
used better (mainly instead of max>0, r>0 is used). Whether to wakeup
is reevaluated every time, instead of somehow doing the wrong thing
and compensating for it with a flag.
This fixes the specific situation when the device buffer is full, and
we don't want to buffer more data. In the old code, this wasn't handled
correctly: the AO went to sleep forever, because it prevented proper
wakeup by the AO driver, and as consequence never asked the core for new
data. Commit 4fa3ffeb was a hack-fix against this, and now that we have
a proper solution, this hack is removed as well.
Also make the refill threshold consistent and always use 1/4 of the
buffer. (The threshold is used for situations when an AO doesn't
support proper waiting or chunked processing.)
This commit will probably cause a bunch of regressions again.
With --gapless-audio=no, changing from one file to the next apparently
made it hang, until the player was woken up by unrelated events like
input. The reason was that the AO doesn't notify the player of EOF
properly. the played was querying ao_eof_reached(), and then just went
to sleep, without anything waking it up.
Make it event-based: the AO wakes up the playloop if the EOF state
changes.
We could have fixed this in a simpler way by synchronously draining the
AO in these cases. But I think proper event handling is preferable.
Fixes: #1069
CC: @mpv-player/stable (perhaps)
The audio/video sync code in player/audio.c calls ao_reset() each time
audio decoding is entered, but the player is paused, and there would be
more than 1 sample to skip to make audio start match with video start.
This caused a wakeup feedback loop with push.c.
CC: @mpv-player/stable
bstr.c doesn't really deserve its own directory, and compat had just
a few files, most of which may as well be in osdep. There isn't really
any justification for these extra directories, so get rid of them.
The compat/libav.h was empty - just delete it. We changed our approach
to API compatibility, and will likely not need it anymore.
The original intention was probably to avoid unnecessarily high numbers
of wakeups. Change it to wait at most 25% of buffer time instead of 75%
until refilling. Might help with the dsound problems in issue #1024, but
I don't know if success is guaranteed.
mp_make_wakeup_pipe() always fails on win32. If this call fails on Linux
(and e.g. ao_alsa is used), this will probably burn CPU since poll()
won't work on the invalid file descriptor, but whatever, the failure
case is obscure enough.
Logic for this was missing from pull.c. For push.c it was missing if the
driver didn't support it. But even if the driver supported it (such as
with ao_alsa), strange behavior was observed by users. See issue #933.
Always check explicitly whether the AO is in paused mode, and if so,
don't drain.
Possibly fixes#933.
CC: @mpv-player/stable
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 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).
wm4
Nicolas F
Rostislav Pehlivanov
Niklas Haas