As ao_pipewire is probed first if a user does not have PipeWire running
they will see a scary warning message even if another AO afterwards is
probed fine.
Tone down the error message so as not to confuse users.
ao-volume is represented in the code with a `struct ao_control_vol_t`
which contains volumes for two channels, left and right.
However the code implementing this property in command.c never treats
these values individually. They are always averaged together.
On the other hand the code in the AOs handling these values also has to
handle the case where *not* exactly two channels are handled.
So let's remove the `struct ao_control_vol_t` and replace it with a
simple float.
This makes the semantics clear to AO authors and allows us to drop some code from the AOs and command.c.
In debug mode the macro causes an assertion failure.
In release mode it works differently and tells the compiler that it can
assume the codepath will never execute. For this reason I was conversative
in replacing it, e.g. in mpv-internal code that exhausts all valid values
of an enum or when a condition is clear from directly preceding code.
[motivation]
Seeking on MacOS appears to be lagged when users connect
to wireless audio output (airpods for example).
This commit attempts to fixmpv-player/mpv#10270
[observation]
1. When using other media player (VLC to be exact) simultaneously,
the lagging on seek disappear. We could guess that the AudioDevice
is on some sort of "warm-up" state.
See mpv-player/mpv#9243 for detailed description.
2. `AudioOutputUnitStart` takes significant longer time after each seek
or pause/play when using wireless output devices compares to wired devices.
[rationale]
After investigate codes in ao_coreaudio.c, it appears that the the `stop`
function was used as `ao_driver.reset` function. Therefore every seek
and pause would call `AudioOutputUnitStop`.
It turns out that `ao_driver.reset` function is used in `ao_reset`.
And `ao_reset` function is used to clean up the state of current `ao`
so I think `AudioUnitReset` is more proper than `AudioOutputUnitStop`
under this semantics.
Since ao_coreaudio use pull base mechanism, audio playback behaviors
upon pause/seek could be handled by callback function
(streaming silence when paused) so there is no need to stop AudioUnit when resetting.
Therefore using `AudioUnitReset` as `ao_driver.reset` looks proper.
Additionally, after using proper reset, the AudioUnit that represents
hardware I/O devices doesn't need to be restart everytime seek/pause actions happen.
Restarting wireless devices simply takes longer in MacOS which is
the root cause of lagging observed by users when they seek or pause/play media.
[method]
Use `AudioUnitReset` for ao_driver.reset.
When a pull AO reaches reaches EOF then ao_read_data() will set
p->playing = false.
Because the ao is marked as not playing ao_set_pause(true) will not
reset the AO.
This keeps the output stream unintentionally open.
Fixes#9835
This allows us to more easily see the datapath from mpv to pipewire.
We know how often the callbacks are triggered, how big the buffers are
and how much data mpv provides to pipewire.
This allows the core of mpv to know about issues in the AO.
Otherwise playback will just freeze as no more data callbacks are sent
by PipeWire.
Also it allows mpv to try to reconnect the AO or find another, working
AO.
We want to add more logic to the stream event handler.
This logic should not be triggered during normal stream shutdown, so we
remove the listener beforehand.
The AO is feature-complete now.
As PipeWire also provides compatibility with PulseAudio, ALSA and Jack
we should put it before those for the autodetection to work.
The pure presence of PipeWire does not mean that it is actually driving
the audio session. For example it could only be meant for video.
Currently there is no proper API to detect this (see [0]), so we check
for the presence of audio sinks.
As soon as a proper API exists, we should use that.
[0] https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/1835
Specifying the id of the target node during stream connect is
deprecated. Instead the property target.object should be used to link
by target serial or name. Using the name allows us to drop a bunch of
custom code.
Previously we would only call list_devs() on available AOs if an AO
*did not* have a hotplug_init() callback or for the first one that *did*
have it.
This is problematic when multiple fully functional hotplug-capable AOs
are available.
The second one would not be able to contribute discovered devices.
This problem prevents ao_pipewire from introducing full hotplug support
with hotplug_init().
When a platform has multiple valid AOs that can provide hotplug events
we should try to use the one that also provides playback.
Concretely this will help when introducing hotplug support for
ao_pipewire.
Currently ao_pulse is probed by ao_hotplug_get_device_list() before
ao_pipewire and on the common setups where both AOs could work pulse
will be selected for hotplug handling.
This means that hotplug_init() of ao_pipewire will never be called and
list_devs() has to do its own initialization.
But if ao_pulse is non-functional or not compiled-in suddenly
ao_pipewire *must* implement hotplug_init() for hotplugging events to
work for all.
Also if the hotplug ao_pulse connects to a PulseAudio instance that is
not emulated by the same PipeWire instance as the playback ao_pipewire
the hotplug events are useless.
uau did some investigation and noticed that we do not send a wakeup
event when we encounter end-of-stream in ao_read_data(), in contrast to
the equivalent logic for push AOs in ao_play_data().
Inserting that wakeup fixes the original problem of lack of
reinitialization on a format change without the problems we saw with
the previous attempted fix.
Fixes#10566
The error description in #10545 could indicate that we are overflowing
we are corrupting the buffer metadata ourselves through out-of-bound
writes.
This check is also present in pw-cat so it seems to be expected for
b->requested to exceed the actual available buffer space.
Potential fix for #10545
pw_core_disconnect frees the core, so accessing it afterward to
destroy the context is not allowed.
Instead, just destroy the context, the first thing it does is disconnect
all cores for us.
mpv only remembers volume for two channels.
Always apply the same volume to all channels in case of
non-stereo layout similarly to ao_pulse.
Don't try to do anything smart when averaging volumes,
normally they are equal anyway.
Pass channel volumes to `pw_stream_set_control` as array.
This is correct calling conventions and prevents
right channel muting every time ao-volume property is changed.
Terminate `pw_stream_set_control` calls with 0.
Changes:
* fixed hangups in the loop function and in some other cases
* refactoring according to @michaelforney's recommendations in #8314
* a few minor and/or cosmetic changes
* ability to build ao_sndio using meson
Changes:
- rewrite to use new internal MPV API;
- code refactoring;
- fix buffers size calculations;
- buffer set to auto;
- reset() - clean/reinit device only after errors;
The AO provides a way for mpv to directly submit audio to the PipeWire
audio server.
Doing this directly instead of going through the various compatibility
layers provided by PipeWire has the following advantages:
* It reduces complexity of going through the compatibility layers
* It allows a richer integration between mpv and PipeWire
(for example for metadata)
* Some users report issues with the compatibility layers that to not
occur with the native AO
For now the AO is ordered after all the other relevant AOs, so it will
most probably not be picked up by default.
This is for the following reasons:
* Currently it is not possible to detect if the PipeWire daemon that mpv
connects to is actually driving the system audio.
(https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/1835)
* It gives the AO time to stabilize before it is used by everyone.
Based-on-patch-by: Oschowa <oschowa@web.de>
Based-on-patch-by: Andreas Kempf <aakempf@gmail.com>
Helped-by: Ivan <etircopyhdot@gmail.com>
Ever instance of m_obj_list is a constant and for all of them, the field
is true. Just remove the field all together.
Signed-off-by: Emil Velikov <emil.l.velikov@gmail.com>
This fixes a mismatch between configure working and build time
failing with Linux + OSSv4, enabling compilation on Debian based
Linux systems with the oss4-dev package.
Fixes#9378
Changes:
- code refactored;
- mixer options removed;
- new mpv sound API used;
- add sound devices detect (mpv --audio-device=help will show all available devices);
- only OSSv4 supported now;
Tested on FreeBSD 12.2 amd64.
This makes the behavior of all control messages consistent,
fixing an inconsistency that has been with us since
4d8266c739 - which is the initial
rework of the polyaudio AO into the pulseaudio AO.
Muting the stream also directly triggers an update to the OSD.
When not waiting for the command completion this read of the mute
property may read the old state. A stale read.
Note that this somehow was not triggered on native Pulseaudio, but it is
an issue on Pipewire.
See https://gitlab.freedesktop.org/pipewire/pipewire/-/issues/868
Set pcm state to SND_PCM_STATE_XRUN in case -EPIPE is received,
and handle this state as per the usual logic.
This way snd_pcm_prepare gets called, and the loop continued.
Inspired by a patch posted by malc_ on #mpv.
--audio-stream-silence is a shitty feature compensating for awful
consumer garbage, that mutes PCM at first to check whether it's
compressed audio, using formats advocated and owned by malicious patent
troll companies (who spend more money on their lawyers than paying any
technicians), wrapped in a wasteful way to make it constant bitrate
using a standard whose text is not freely available, and only rude users
want it. This feature has been carelessly broken, because it's
complicated and stupid. What would Jesus do? If not getting an aneurysm,
or pushing over tables with expensive A/V receivers on top of them, he'd
probably fix the feature. So let's take inspiration from Jesus Christ
himself, and do something as dumb as wasting some of our limited
lifetime on this incredibly stupid fucking shit.
This is tricky, because state changes like end-of-audio are supposed to
be driven by the AO driver, while playing silence precludes this. But it
seems code paths for "untimed" AOs can be reused.
But there are still problems. For example, underruns will just happen
normally (and stop audio streaming), because we don't have a separate
heuristic to check whether the buffer is "low enough" (as a consequence
of a network stall, but before the audio output itself underruns).
Create a central function which pumps data through the filter. This also
might fix bogus use of the filter API on flushing. (The filter is just
used for convenience, but I guess the overall result is still simpler.)
It is now the AO's responsibility to handle period size alignment. The
ao->period_size alignment field is unused as of the recent audio
refactor commit. Remove it.
It turns out that ao_alsa shows extremely inefficient behavior as a
consequence of the removal of period size aligned writes in the
mentioned refactor commit. This is because it could get into a state
where it repeatedly wrote single samples (as small as 1 sample), and
starved the rest of the player as a consequence. Too bad. Explicitly
align the size in ao_alsa. Other AOs, which need this, should do the
same.
One reason why it broke so badly with ao_alsa was that it retried the
write() even if all reported space could be written. So stop doing that
too. Retry the write only if we somehow wrote less.
I'm not sure about ao_pulse.
This replaces the two buffers (ao_chain.ao_buffer in the core, and
buffer_state.buffers in the AO) with a single queue. Instead of having a
byte based buffer, the queue is simply a list of audio frames, as output
by the decoder. This should make dataflow simpler and reduce copying.
It also attempts to simplify fill_audio_out_buffers(), the function I
always hated most, because it's full of subtle and buggy logic.
Unfortunately, I got assaulted by corner cases, dumb features (attempt
at seamless looping, really?), and other crap, so it got pretty
complicated again. fill_audio_out_buffers() is still full of subtle and
buggy logic. Maybe it got worse. On the other hand, maybe there really
is some progress. Who knows.
Originally, the data flow parts was meant to be in f_output_chain, but
due to tricky interactions with the playloop code, it's now in the dummy
filter in audio.c.
At least this improves the way the audio PTS is passed to the encoder in
encoding mode. Now it attempts to pass frames directly, along with the
pts, which should minimize timestamp problems. But to be honest, encoder
mode is one big kludge that shouldn't exist in this way.
This commit should be considered pre-alpha code. There are lots of bugs
still hiding.
Previously get_state() would keep setting the cork status
while paused, but it only does for that after underflows now.
Correct this oversight by creating the stream corked for start()
to uncork it at a later time.
fixes#8026
FFmpeg expects those fields to be set on the AVFrame when
encoding audio, not doing so will cause the avcodec_send_frame
call to return EINVAL (at least in recent builds).
When get_state() corks the stream after an underrun happens
priv->playing is incorrectly reset to true, which can cause the
player to miss the underrun entirely. Stop resetting priv->playing
during corking (but not uncorking) to fix this.
The underflow callback introduced in d27ad96 can be called
when the buffer is still full, causing playback to never
resume afterwards since get_state() reports free_samples == 0.
Fix this by fully resetting on underrun, which flushes
the stream and ensures free buffer space.
fixes#7874
The pull mode APIs were previously required to have thread-safe
ao_controls. However, locks were added in b83bdd1 for parity with push
mode. This introduced deadlocks in ao_wasapi.
Instead, only lock ao_control for the push mode APIs.
fixes#7787
See also #7832, #7811. We'll wait for feedback to see if those should
also be closed.
AOs which use the "push" API must set this field now. Actually, this was
sort of always required, but happened to work anyway. The future
intention is to use device_buffer as the pre-buffer amount, which has to
be available right before audio playback is started. "Pull" AOs really
need this too conceptually, just that the API is underspecified.
From what I can see, only ao_null did not do this yet.
Previously, device_buffer defaulted to 0 on pulse. This meant that
commit baa7b5c would always wait with a timeout of 0, leading to
high CPU usage for PulseAudio users.
By setting device_buffer to the number of samples per channel that
PulseAudio sets as its target, this commit fixes this behaviour.
The playback thread may obviously still fill the AO'S entire audio
buffer, which means it unset p->draining, which makes no sense and broke
ao_drain(). So just don't unset it here.
Not sure if this really fixes this, it was hard to reproduce. Regression
due to the recent changes. There are probably many more bugs like this.
Stupid asynchronous nightmare state machine. Give me a language that
supports formal verification (in presence of concurrency) or something.
I feel like this makes slightly more sense. At least it doesn't include
the potentially arbitrary constant latency that is generally included in
the delay value. Also, the buffer status doesn't matter - either we've
filled the entire buffer (then we can wait this long), or there's not
enough data anyway (then the core will wake up the thread if new data is
available).
But ultimately, we have to guess, unless the AO does notify us with
ao_wakeup_playthread().
Draining may now wait for no reason up to 1/4th of the total buffer
time. Shouldn't be a disimprovement in practice.
It's conceivable that ao->driver->reset() will make the audio API wait
for ao_read_data() (i.e. its audio callback) to return. Since we
recently moved the reset() call inside the same lock that ao_read_data()
acquires, this could deadlock. Whether this really happens depends on
how exactly the AO behaves. For example, ao_wasapi does not have this
problem. "Push" AOs are not affected either.
Fix by moving it outside of the lock. Assume ao->driver->start() will
not have this problem.
Could affect ao_sdl, ao_coreaudio (and similar rotten fruit AOs). I'm
unsure whether anyone experienced the problem in practice.
Instead of the relatively subtle underflow handling, simply signal
whether the stream is in a playing state. Should make it more robust.
Should affect ao_alsa and ao_pulse only (and ao_openal, but it's
broken).
For ao_pulse, I'm just guessing. How the hell do you query whether a
stream is playing? Who knows. Seems to work, judging from very
superficial testing.
Just a detail. If wrong (not unlikely because I'm just guessing my own
messy state machine), this will make the player freeze due to waiting
for something that never happens. Enjoy.
The feeder thread basically woke up the core and itself too often, and
caused some CPU overhead. This was caused by the recent buffer.c
changes.
For one, do not let ao_read_data() wake up the core, and instead rely on
the feeder thread's own buffer management. This is a bit strange, since
the change intended to unify the buffer management, but being more
consequent about it is better deferred to later, when the buffer
management changes again anyway. And also, the "more" condition in the
feeder thread seems outdated, or at least what made it make sense has
been destroyed, so do something that may or may not be better. In any
case, I'm still not getting underruns with ao_alsa, but the wakeup
hammering is gone.
This affects "pull" AOs only: ao_alsa, ao_pulse, ao_openal, ao_pcm,
ao_lavc. There are changes to the other AOs too, but that's only about
renaming ao_driver.resume to ao_driver.start.
ao_openal is broken because I didn't manage to fix it, so it exits with
an error message. If you want it, why don't _you_ put effort into it? I
see no reason to waste my own precious lifetime over this (I realize the
irony).
ao_alsa loses the poll() mechanism, but it was mostly broken and didn't
really do what it was supposed to. There doesn't seem to be anything in
the ALSA API to watch the playback status without polling (unless you
want to use raw UNIX signals).
No idea if ao_pulse is correct, or whether it's subtly broken now. There
is no documentation, so I can't tell what is correct, without reverse
engineering the whole project. I recommend using ALSA.
This was supposed to be just a simple fix, but somehow it expanded scope
like a train wreck. Very high chance of regressions, but probably only
for the AOs listed above. The rest you can figure out from reading the
diff.
wasapi/coreaudio/sdl were affected, alsa/pusle were not.
The confusion here was that resume() has different meaning with pull and
push AOs.
Fixes: #7772
Regression since the recent refactor. How did nobody notice?
This happened because the push code now calls the function for the pull
code. Both the former and latter apply the volume, so oops.
The recent change to the common code removed all calls to ->drain. It's
currently emulated via a timed sleep and polling ao_eof_reached(). That
is actually fallback code for AOs which lacked draining. I could just
readd the drain call, but it was a bad idea anyway. My plan to handle
this better is to require the AO to signal a underrun, even if
AOPLAY_FINAL_CHUNK is not set. Also reinstate not possibly waiting for
ao_lavc.c. ao_pcm.c did not have anything to handle this; whatever.
This is preparation to further cleanups (and eventually actual
improvements) of the audio output code.
AOs are split into two classes: pull and push. Pull AOs let an audio
callback of the native audio API read from a ring buffer. Push AOs
expose a function that works similar to write(), and for which we start
a "feeder" thread. It seems making this split was beneficial, because of
the different data flow, and emulating the one or other in the AOs
directly would have created code duplication (all the "pull" AOs had
their own ring buffer implementation before it was cleaned up).
Unfortunately, both types had completely separate implementations (in
pull.c and push.c). The idea was that little can be shared anyway. But
that's very annoying now, because I want to change the API between AO
and player.
This commit attempts to merge them. I've moved everything from push.c to
pull.c, the trivial entrypoints from ao.c to pull.c, and attempted to
reconcile the differences. It's a mess, but at least there's only one
ring buffer within the AO code now. Everything should work mostly the
same. Pull AOs now always copy the audio data under a lock; before this
commit, all ring buffer access was lock-free (except for the decoder
wakeup callback, which acquired a mutex). In theory, this is "bad", and
people obsessed with lock-free stuff will hate me, but in practice
probably won't matter. The planned change will probably remove this
copying-under-lock again, but who knows when this will happen.
One change for the push AOs now makes it drop audio, where before only a
warning was logged. This is only in case of AOs or drivers which exhibit
unexpected (and now unsupported) behavior.
This is a risky change. Although it's completely trivial conceptually,
there are too many special cases. In addition, I barely tested it, and
I've messed with it in a half-motivated state over a longer time, barely
making any progress, and finishing it under a rush when I already should
have been asleep. Most things seem to work, and I made superficial tests
with alsa, sdl, and encode mode. This should cover most things, but
there are a lot of tricky things that received no coverage. All this
text means you should be prepared to roll back to an older commit and
report your problem.
Ancient Linux audio output. Apparently it survived until now, because
some BSDs (but not all) had use of this. But these should work with
ao_sdl or ao_openal too (that's why these AOs exist after all). ao_oss
itself has the problem that it's virtually unmaintainable from my point
of view due to all the subtle (or non-subtle) difference. Look at the
ifdef mess and the multiple code paths (that shouldn't exist) in the
removed source code.
I wonder what this even is. I've never heard of anyone using it, and
can't find a corresponding library that actually builds with it. Good
enough to remove.
It was always marked as "experimental", and had inherent problems that
were never fixed. It was disabled by default, and I don't think anyone
is using it.
Looks like the recent change to this actually made it crash whenever
audio happened to be initialized first, due to not setting the
mux_stream field before the on_ready callback. Mess a way around this.
Also remove a stray unused variable from ao_lavc.c.
In shared mode, we previously tried to feed the full native format to
IsFormatSupported in the hopes that the "closest match" returned was
actually that.
Turns out, IsFormatSupported will always return the mix format if we
don't use the mix format's sample rate. This will also clobber our
choice of channel map with the mix format channel map even if our
desired channel map is supported due to surround emulation.
The solution is to not bother trying to use anything other than the mix
format sample rate. While we're at it, we might as well use the mix
format PCM sample format (always float32) since this conversion will
happen anyway and may avoid unecessary dithering to intermediate
integer formats if we are already resampling or channel mixing.
Change all OPT_* macros such that they don't define the entire m_option
initializer, and instead expand only to a part of it, which sets certain
fields. This requires changing almost every option declaration, because
they all use these macros. A declaration now always starts with
{"name", ...
followed by designated initializers only (possibly wrapped in macros).
The OPT_* macros now initialize the .offset and .type fields only,
sometimes also .priv and others.
I think this change makes the option macros less tricky. The old code
had to stuff everything into macro arguments (and attempted to allow
setting arbitrary fields by letting the user pass designated
initializers in the vararg parts). Some of this was made messy due to
C99 and C11 not allowing 0-sized varargs with ',' removal. It's also
possible that this change is pointless, other than cosmetic preferences.
Not too happy about some things. For example, the OPT_CHOICE()
indentation I applied looks a bit ugly.
Much of this change was done with regex search&replace, but some places
required manual editing. In particular, code in "obscure" areas (which I
didn't include in compilation) might be broken now.
In wayland_common.c the author of some option declarations confused the
flags parameter with the default value (though the default value was
also properly set below). I fixed this with this change.
This seems to be an older bug. It set priv->outputfilename to a new
talloc-allocated string, but the field is also managed as string option,
so talloc will free it first, then m_option_free() is called on the
dangling pointer. Possibly this is caused by the earlier ta destruction
order change.
Before this commit, option declarations used M_OPT_MIN/M_OPT_MAX (and
some other identifiers based on these) to signal whether an option had
min/max values. Remove these flags, and make it use a range implicitly
on the condition if min<max is true.
This requires care in all cases when only M_OPT_MIN or M_OPT_MAX were
set (instead of both). Generally, the commit replaces all these
instances with using DBL_MAX/DBL_MIN for the "unset" part of the range.
This also happens to fix some cases where you could pass over-large
values to integer options, which were silently truncated, but now cause
an error.
This commit has some higher potential for regressions.
Move the "old" mostly command line parsing and option management related
code to m_config_frontend.c/h. Move the the code that enables other part
of the player to access options to m_config_core.c/h. "frontend" is out
of lack of creativity for a better name.
Unfortunately, the separation isn't quite clean yet. m_config_frontend.c
still references some m_config_core.c implementation details, and
m_config_new() is even left in m_config_core.c for now. There some odd
functions that should be removed as well (marked as "Bad functions").
Fixing these things requires more changes and will be done separately.
struct m_config is left with the current name to reduce diff noise.
Also, since there are a _lot_ source files that include m_config.h, add
a replacement m_config.h that "redirects" to m_config_core.h.
A previous commit moved the underrun reporting to report_underruns(),
and called it from get_space(). One reason was that I worried about
printing a log message from a "realtime" callback, so I tried to move it
out of the way. (Though there's little justification other than a bad
feeling. While an older version of the pull code tried to avoid any
mutexes at all in the callback to accommodate "requirements" from APIs
like jackaudio, we gave up on that. Nobody has complained yet.)
Simplify this and move underrun reporting back to the callback. But
instead of printing the message from there, move the message into the
playloop. Change the message slightly, because ao->log is inaccessible,
and without the log prefix (e.g. "[ao/alsa]"), some context is missing.
AOs can report audio underruns, but only ao_alsa and ao_sdl (???)
currently do so. If the AO was marked as not reporting it, the cache
state was used to determine whether playback was interrupted due to slow
input.
This caused problems in some cases, such as video with very low video
frame rate: when a new frame is displayed, a new frame has to be
decoded, and since there it's so much further into the file (long frame
durations), the cache gets into an underrun state for a short moment,
even though both audio and video are playing fine. Enlarging the audio
buffer didn't help.
Fix this by making all AOs report underruns. If the AO driver does not
report underruns, fall back to using the buffer state.
pull.c behavior is slightly changed. Pull AOs are normally intended to
be used by pseudo-realtime audio APIs that fetch an audio buffer from
the API user via callback. I think it makes no sense to consider a
buffer underflow not an underrun in any situation, since we return
silence to the reader. (OK, maybe the reader could check the return
value? But let's not go there as long as there's no implementation.)
Remove the flag from ao_sdl.c, since it just worked via the generic
mechanism. Make the redundant underrun message verbose only.
push.c seems to log a redundant underflow message when resuming (because
somehow ao_play_data() is called when there's still no new data in the
buffer). But since ao_alsa does its own underrun reporting, and I only
use ao_alsa, I don't really care.
Also in all my tests, there seemed to be a rather high delay until the
underflow was logged (with audio only). I have no idea why this happened
and didn't try to debug this, but there's probably something wrong
somewhere.
This commit may cause random regressions.
See: #7440
If ao_add_events() is used, but all events flags are already set, then
we don't need to wakeup the core again.
Also, make the underrun message "exact" by avoiding the race condition
mentioned in the comment.
Avoiding redundant wakeups is not really worth the trouble, and it's
actually just a bonus in the change making the ao_underrun_event()
function return whether a new underrun was set, which is needed by the
following commit.
Before this commit, runtime changes were only applied if something else
caused audio to be reinitialized. Now setting them reinitializes audio
explicitly.
The code is very basic:
- only handles gamepads, could be extended for generic joysticks in the
future.
- only has button mappings for controllers natively supported by SDL2.
I heard more can be added through env vars, there's also ways to load
mappings from text files, but I'd rather not go there yet. Common ones
like Dualshock are supported natively.
- analog buttons (TRIGGER and AXIS) are mapped to discrete buttons using an
activation threshold.
- only supports one gamepad at a time. the feature is intented to use
gamepads as evolved remote controls, not play multiplayer games in mpv :)
This was all dead code. Commit 995c47da9a (over 3 years ago) removed all
uses of the controls.
It would be nice if AOs could apply a linear gain volume, that only
affects the AO's audio stream for low-latency volume adjust and muting.
AOCONTROL_HAS_SOFT_VOLUME was supposed to signal this, but to use it,
we'd have to thoroughly check whether it really uses the expected
semantics, so there's really nothing useful left in this old code.
See previous commits. ao_sdl is worthless, but it might be a good test
for pull-based AOs.
This stops using the old underrun reporting if the new one is enabled.
Also, since the AO's behavior can in theory not be according to
expectations, this needs to be enabled for every single pull AO
separately.
For some reason, in certain cases I get multiple underrun warnings while
cache-pausing is active. It fills the cache, restarts the AO,
immediately underruns again, and then fills the cache again. I'm not
sure why this happens; maybe ao_sdl tries to catch up when it shouldn't.
Who knows.
I think this was _always_ wrong. Due to the line above the first changed
line, buffered_bytes==bytes always. I can only hope I broke this in a
less under-tested edit when I originally wrote this.
Fixes: c5a82f729b
AOs can now call ao_underrun_event() (in any context) if an underrun has
happened. It will print a message.
This will be used in the following commits. But for now, audio.c only
clears the underrun bit, so that subsequent underruns still print the
warning message.
Since the underrun flag will be used in fragile ways by the playback
state machine, there is the "reports_underruns" field that signals
strong support for underrun reporting. (Otherwise, underrun events will
not be used by it.)
This commit tries to prepare for better underrun reporting. The goal is
to report underruns relatively immediately. Until now, this happened
only when play() was called. Change this, and abuse that get_delay() is
called "relatively often" - this reports the underrun immediately in
practice.
Background:
In commit 81e51a15f7 (and also e38b0b245e), we were quite confused
about ALSA underrun handling. The commit message showed uncertainty how
case 3 happened, but it's blindingly obvious and simple.
Actually reading the code shows that ALSA does not have a concept of a
"final chunk" (or we don't use it). It's obvious we never pass the
AOPLAY_FINAL_CHUNK flag along to the ALSA API in any way. The only thing
we do is simply writing a partial fragment. Of course this will cause an
underrun. Doing a partial write saves us the trouble to pad the last
frame with silence, or so.
The main reason why the underrun message was avoided was that play() was
never called with a non-0 sample count again (except if reset() was
called before that). That was OK, at least the goal of avoiding the
unwanted message was reached. (And the original "bogus" message at end
of playback was perfectly correct, as far as ALSA goes.)
If network stalls, play() will called again only once new data is
available. Obviously, this could take a long time, thus it's too late.
It turns out that case 2) mentioned in the previous commit happened
quite often when playback ended normally.
There is probably a legitimate underrun with normal buffer sizes (100
ms, 4 fragments, gapless audio in "weak" mode). This is a result of the
player waiting for video to end, and/or the time needed to kill the
video window. The former case means that it depends on your test case
whether it happens (a file where video ends slightly before audio is
less likely to trigger it).
This in turn is due to how gapless playback works. Achieving not having
a "gap" requires queuing the audio of the next file without playing a
partial chunk (as AOPLAY_FINAL_CHUNK would do). The partial chunk is
then played as part of the first chunk played from the next file. But if
it detects "later" that there is no next file, it still needs to get rid
of the last fragment with AOPLAY_FINAL_CHUNK. At this point it's too
late, and an underrun may have actually happened. The way the player
uninits and reinits the entire playback engine for the next file in a
"serial" manner means it cannot know in advance whether this works.
This is the reason why the idiot who added the underrun exception for
the last chunk in play() was wrong (I wrote that btw., before you accuse
me of being rude). Yes, it's a real underrun, and you could probably
hear it.
This XRUN (aka underrun) message was printed in the following
situations:
1) legitimate underrun during playback
2) legitimate underrun when playing final chunk
3) bogus underrun when playing final chunk
The old underrun case (in play()) happens in cases 1) and 2) as well,
but 3) did not happen. It appears 3) is indeed something that happens,
although it's not known for sure. It's still pretty annoying, so remove
the new XRUN message.
When testing, care should be taken to play with buffer sizes, video
versus no video, and gapless enabled/disabled. Also, suspending the
player with Ctrl+Z in the terminal (SIGSTOP) and then resuming is a good
way to trigger a "normal" underrun.
ioctl(..., SNDCTL_DSP_CHANNELS, &nchannels) for not supported
nchannels does not return an error and instead set nchannels to
the default value.
Instead of failing with no audio, fallback to stereo.
This flag makes mpv continue using the PulseAudio driver even if the
sink is suspended.
This can be useful if JACK is running with PulseAudio in bridge mode and
the sink-input assigned to mpv is the one JACK controls, thus being
suspended.
By forcing mpv to still use PulseAudio in this case, the user can now
adjust the sink to an unsuspended one.
According to ALSA doxy, EPIPE is a synonym to SND_PCM_STATE_XRUN,
and that is a state that we should attempt to automatically recover
from. In case recovery fails, log an error and return zero.
A warning message will still be output for each XRUN since those
are not something we should generally be receiving.
This has been way too long coming, and for me to notice that a
whole lot of ao_alsa functions do an early return if the AO is
paused.
For the STATE_SETUP case, I had this reproduced once, and never
since. Still, seems like we can start calling this function before
the ALSA device has been fully initialized so we might as well
early exit in that case.
ao->device_buffer will only affect the enqueue size if the latter
is not specified. In other word, its intended purpose will solely
be setting/guarding the soft buffer size.
This guarantees that the soft buffer size will be consistent no
matter a specific enqueue size is set or not. (In the past it
would drop to the default of the generic audio-buffer option.)
opensles-frames-per-buffer has been renamed to opensles-frames-per
-enqueue, as it was never purposed to set the soft buffer size. It
will only make sure the size is never smaller than itself, just as
before.
opensles-buffer-size-in-ms is introduced to allow easy tuning of
the relative (i.e. in time) soft buffer size (and enqueue size,
unless the aforementioned option is set). As "device buffer" never
really made sense in this AO, this option OVERRIDES audio-buffer
whenever its value (including the default) is larger than 0.
Setting opensl-buffer-size-in-ms to 1 allows you to equate the soft
buffer size to the absolute enqueue size set with opensl-frames-per
-enqueue conveniently (unless it is less than 1ms).
When both are set to 0, audio-buffer will be the ultimate fallback.
If audio-buffer is also 0, the AO errors out.
Fixes a bug with alsa dmix on Fedora 29. After several minutes,
audio suddenly becomes bad and muted.
Actually, I don't know what causes this. Probably this is a bug in alsa.
In any case, as snd_pcm_status() returns not only 'avail', but also other
fields such as tstamp, htstamp, etc, this could be considered a good
simplification, as only avail is required for this function.
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.
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.
Print them as a warning.
Note that there may be some cases where it underruns, without being a
bad condition. This could possibly happen e.g. if the last chunk is
written, and then it resumes playback some time after that. Eventually I
want to add more code to avoid such spurious warnings.
There is a dedicated thread for feeding audio to the ALSA API from a
buffer with a larger size. There is little reason to have such a large
device buffer.
One can now set the number of buffers and the buffer size.
This can reduce the CPU usage and the total latency stays mostly the same.
As there are sync mechanisms the A/V sync continue intact and working.
It also modifies 6.1 channel order, as per OpenAL spec
and add AOPLAY_FINAL_CHUNK support
OpenAL Soft's AL_SOFT_source_latency extension allows one to correctly
get the device output latency, facilitating the syncronization with
video.
Also added a simpler generic fallback that does not take into account
latency of the device.
Uses OpenAL Soft's AL_DIRECT_CHANNELS_SOFT extension and can be controlled through
a new CLI option, --openal-direct-channels.
This allows one to send the audio data direrctly to the desired channel without
effects applied.
Although half (non-fast track on sink rate) or one-third (non-fast track not on sink rate) of the buffer size of the created AudioTrack instance as the SL Enqueue buffer size is basically enough for dropout-free playback, only using the full size can avoid stutter upon (re)start of playback.
Here are the various buffer sizes on different track/sink rate when on Bluetooth audio on Android O:
aptX @ 48kHz:
Sink rate: 48000 Hz
44100 Hz: 10632 frames (241.09 ms)
48000 Hz: 11544 frames (240.50 ms)
88200 Hz: 21216 frames (240.54 ms)
96000 Hz: 23088 frames (240.50 ms)
176400 Hz: 42384 frames (240.27 ms)
192000 Hz: 46128 frames (240.25 ms)
SBC/AAC/aptX @ 44.1kHz:
Sink rate: 44100 Hz
44100 Hz: 10776 frames (244.35 ms)
48000 Hz: 11748 frames (244.75 ms)
88200 Hz: 21552 frames (244.35 ms)
96000 Hz: 23448 frames (244.25 ms)
176400 Hz: 43056 frames (244.08 ms)
192000 Hz: 46848 frames (244.00 ms)
The above results were produced with the following code:
import android.media.AudioAttributes;
import android.media.AudioFormat;
import android.media.AudioTrack;
class AudioInfo {
public static void main(String[] args) {
int nosr = AudioTrack.getNativeOutputSampleRate(3);
System.out.printf("Sink rate: %d Hz\n", nosr);
int[] rates = {44100,48000,88200,96000,176400,192000};
for (int rate: rates) {
AudioAttributes aa = new AudioAttributes.Builder().setFlags(256).build();
AudioFormat af = new AudioFormat.Builder().setSampleRate(rate).build();
AudioTrack at = new AudioTrack(aa, af, 4, 1, 0);
int sr = at.getSampleRate();
int bs = at.getBufferSizeInFrames();
float ms = bs * (float) 1000 / sr;
at.release();
System.out.printf("%d Hz: %d frames (%.2f ms)\n", sr, bs, ms);
}
}
}
Therefore bumping the device buffer size to 250ms.
If you set desired.samples to 0, SDL will return a default buffer size
on obtained.samples. This was broken, because ceil_power_of_two(0)
returns 1. Since 0 is usually not considered a power of two, this is
probably correct, but we still want to set desired.samples to 0 in this
case.
You can use --audio-buffer=0 to minimize the audio buffer size. But if
the AO reports no device buffer size (like e.g. ao_jack does), then the
buffer size is actually 0, and playback can never work properly.
Make it fallback to a size of 1, which is unlikely to work properly, but
you get what you asked for, instead of a freeze.
While the soft buffer size is already by default 200ms, it is not enough to guarantee dropout-free playback on Bluetooth audio. Bumping the device buffer size to the same value seems to suffice.
