Same as ffmpeg uses. Such big values does not make sense probably, but
let's not overflow values and maybe one day it will be useful.
Fixes signed integer overflow.
Currently, the softvol gain control attempts to clip floating point ao
formats within -1 and +1. However, this is "optimized out" at unity gain,
where no clipping is applied. This results in inconsistent behavior when
the source audio is already out of -1 and +1 range, where a gain of 0.99
results in clipping, but not at exactly 1.
Since a big advantage of floating point audio data is that they do not
lose information through out-of-range data because the ao sink can apply
suitable negative gain to prevent clipping before converting them to
integer formats, clipping should not be performed on these data.
Fix this by removing the existing clipping behavior. It now results in
a simple multiplication, which faciliates compiler auto-vectorization
of this operation over audio data.
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.
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.
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.
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>
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.
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.
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.
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.
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.
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.
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 :)
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.)
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.
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.
stdatomic.h defines no atomic_float typedef. We can't just use _Atomic
unconditionally, because we support compilers without C11 atomics. So
just create a custom atomic_float typedef in the wrapper, which uses
_Atomic in the C11 code path.
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.
This is pretty pointless, but I believe it allows us to claim that the
new code is not affected by the copyright of the old code. This is
needed, because the original mp_audio struct was written by someone who
has disagreed with LGPL relicensing (it was called af_data at the time,
and was defined in af.h).
The "GPL'ed" struct contents that surive are pretty trivial: just the
data pointer, and some metadata like the format, samplerate, etc. - but
at least in this case, any new code would be extremely similar anyway,
and I'm not really sure whether it's OK to claim different copyright. So
what we do is we just use AVFrame (which of course is LGPL with 100%
certainty), and add some accessors around it to adapt it to mpv
conventions.
Also, this gets rid of some annoying conventions of mp_audio, like the
struct fields that require using an accessor to write to them anyway.
For the most part, this change is only dumb replacements of mp_audio
related functions and fields. One minor actual change is that you can't
allocate the new type on the stack anymore.
Some code still uses mp_audio. All audio filter code will be deleted, so
it makes no sense to convert this code. (Audio filters which are LGPL
and which we keep will have to be ported to a new filter infrastructure
anyway.) player/audio.c uses it because it interacts with the old filter
code. push.c has some complex use of mp_audio and mp_audio_buffer, but
this and pull.c will most likely be rewritten to do something else.
I plan to remove the S24 sample formats in mpv. It seems like we should
still support this _somehow_ in AOs though. So the idea is to convert
the data to more obscure representations (that would not be useful for
filtering etc. anyway) within the AO.
This commit adds helper to enable this. ao_convert_fmt is meant to
provide mechanisms for this, rather than a generic audio format
description (as the latter leads only to overly generic misery). The
conversion also supports only cases which we think will be needed at
all.
The main advantage of this approach is that we get S24 out of sight,
and that we could support other crazy formats (like S20). The main
disadvantage is that usually S32 will be selected (if both S32 and S24
are available), and there's no user control to force S24. That doesn't
really matter though, and at worst makes testing harder or will lead
to unpleasant arguments with audiophiles (they'd be wrong anyway).
ao_convert_fmt.pad_lsb is ignored, although if we ever find a case in
which playing S32 with data in the LSBs breaks when playing it as padded
24 bit format. (For example, WAVEFORMATEXTENSIBLE recommends setting the
unused bits to 0 if wValidBitsPerSample implies LSB padding.)
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.
All contributors of the current code have agreed. ao.c requires a
"driver" entry for each audio output - we assume that if someone who
didn't agree to LGPL added a line, it's fine for ao.c to be LGPL
anyway. If the affected audio output is not disabled at compilation
time, the resulting binary will be GPL anyway, and ootherwise the
code is not included.
The audio output code itself was inspired or partially copied from
libao in 7a2eec4b59 (thus why MPlayer's audio code is named libao2).
Just to be sure we got permission from Aaron Holtzman, Jack Moffitt, and
Stan Seibert, who according to libao's SVN history and README are the
initial author. (Something similar was done for libvo, although the
commit relicensing it forgot to mention it.)
242aa6ebd40: anders mostly disagreed with the LGPL relicensing, but we
got permission for this particular commit.
0ef8e555735: nick could not be reached, but the include statement was
removed again anyway.
879e05a7c17: iive agreed to LGPL v3+ only, but this line of code was
removed anyway, so ao_null.c can be LGPL v2.1+.
9dd8f241ac2: patch author could not be reached, but the corresponding
code (old slave mode interface) was completely removed later.
For example, previously, --audio-device='alsa/' would provide ao->device="" to
the alsa driver in spite of the fact that this is an already parsed option. To
avoid requiring a check of ao->device[0] in every driver, make sure this never
happens.
Long planned. Leads to some sanity.
There still are some rather gross things. Especially g_groups is ugly,
and a hack that can hopefully be removed. (There is a plan for it, but
whether it's implemented depends on how much energy is left.)
This will make it easier for AOs to add explicit default device entries.
(See next commit.)
Hopefully this change doesn't lead accidentally to bogus "Default"
entries to appear, but then it can only happen if the device ID is
empty, which would mean the underlying audio API returned bogus entries.
Use the device name as fallback. This is ugly, but still better than
skipping the description entirely. This can be an issue on ALSA, where
the API can return entries without proper description.
Kacper Michajłow
Misaki Kasumi
nanahi
Thomas Weißschuh
sfan5
rim
Emil Velikov
wm4
Aman Gupta
Stefano Pigozzi
Kevin Mitchell