Commit Graph

81 Commits

Author SHA1 Message Date
sfan5 aa300f8023 ao/pulse: fix incorrect state reported after reset
fixes #8768
2021-04-29 17:06:29 +02:00
Thomas Weißschuh 63d71ba4ec ao/pulse: signal the mainloop when ops are done
Without the explicit signal the call to pa_threaded_mainloop_wait()
will not return as soon as possible.

Fixes 4f07607888
See #8633
2021-03-11 23:37:13 +02:00
Thomas Weißschuh 4f07607888 ao/pulse: wait for command completion when setting volume or mute
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
2021-03-09 23:08:16 +02:00
sfan5 fb736b49f1 ao/pulse: create the stream corked
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
2020-08-26 16:14:29 +02:00
sfan5 7a7a0a78b5 ao/pulse: fix reporting of playing state
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.
2020-07-12 23:44:41 +02:00
sfan5 f3b29a680c ao/pulse: flush stream on underrun
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
2020-07-12 23:34:08 +02:00
Nicolas F 0fb02f181f ao/pulse: properly set device_buffer
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.
2020-06-07 22:16:49 +03:00
wm4 08b198aab4 audio: further simplify internal audio API somewhat
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.
2020-06-02 20:43:49 +02:00
wm4 d27ad96542 audio: redo internal AO API
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.
2020-06-01 01:08:16 +02:00
wm4 9885952c2a audio: remove ao_driver.drain
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.
2020-05-27 21:04:32 +02:00
wm4 26f4f18c06 options: change option macros and all option declarations
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.
2020-03-18 19:52:01 +01:00
wm4 cde94e83a9 audio/out: rip out old unused app/softvolume reporting
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.
2019-10-11 21:05:11 +02:00
Jan Ekström 69e4a5772a ao_pulse: add the newly added mappings for TrueHD/DTS-HD formats
Originally DTS-HD was mapped to PA_ENCODING_DTS_IEC61937 which I'm
actually not sure if it ever worked.
2019-09-27 00:23:36 +03:00
Térence Clastres 41f4e8d73a ao_pulse: add --pulse-allow-suspended
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.
2019-09-21 12:54:36 +02:00
Tom Yan 9d6b15ab32 ao_pulse: fix tlength calculation
also remove the now unused non-sensical af_fmt_seconds_to_bytes.
2018-09-01 16:14:11 +02:00
wm4 66810c1550 ao_pulse: reduce requested device buffer size
Same deal as with the previous commit for ALSA.

Untested.
2018-04-15 23:11:33 +03:00
Niklas Haas bbe8bb0ae9
ao_pulse: reorder format choice
Right now, the current order pretty much means that pulse defaults to
S16 for arbitrary unsupported formats, but fallback to float would make
more sense since it's the easiest to convert everything to without
requiring dithering, and PA will probably just internally convert things
to float anyway.

Also move S32 above S16, which essentially means format_maps is sorted
by preference. (Although ao_pulse currently ignores this and always
picks the first as a fallback)
2017-06-23 21:12:44 +02:00
wm4 04df16bfd3 ao_pulse, ao_rsound: change license to LGPL
All authors have agreed.

One exception is 71247a97b3, whose author was not asked, but we deem
the change as trivial. (And technically it was replaced when the audio
chain dropped non-native endian sample formats.)
2017-05-08 14:09:49 +02:00
wm4 809d160c1e options: remove remaining deprecated audio device selection options 2017-04-23 17:51:55 +02:00
wm4 1a2319f3e4 options: remove deprecated sub-option handling for --vo and --ao
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.)
2016-11-25 21:17:25 +01:00
wm4 13786dc643 audio/out: deprecate device sub-options
We have --audio-device, which can force the device. Also add something
describing to this extent to the manpage.
2016-09-05 21:26:39 +02:00
wm4 69283bc0f8 options: deprecate suboptions for the remaining AO/VOs 2016-09-05 21:26:39 +02:00
wm4 d191d76e52 ao_pulse: fix some volume control rounding issues
Volume could get easily "stuck" or making too huge steps when doing
things like "add ao-volume 1".
2016-07-14 18:11:14 +02:00
wm4 c1002f6a28 ao_pulse: attempt to fall back to an arbitrary sample format
Normally, PulseAudio accepts any combination of sample format, sample
rate, channel count/map. Sometimes it does not. For example, the channel
rate or channel count have fixed maximum values. We should not fail
fatally in such cases, but attempt to fall back to a working format.

We could just send pass an "unset" format to Pulse, but this is not too
attractive. Pulse could use a format which we do not support, and also
doing so much for an obscure corner case is not reasonable. So just pick
a format that is very likely supported.

This still could fail at runtime (the stream could fail instead of going
to the ready state), but this sounds also too complicated. In
particular, it doesn't look like pulse will tell us the cause of the
stream failure. (Or maybe it does - but I didn't find anything.)

Last but not least, our fallback could be less dumb, and e.g. try to fix
only one of samplerate or channel count first to reduce the loss, but
this is also not particularly worthy the effort.

Fixes #2654.
2016-01-05 19:52:05 +01:00
wm4 861c126b08 ao_pulse: check for sample rate bounds
pa_format_info_valid() does not do this. (Although there is a proposed
patch on the PulseAudio mailing list.)

See #2654.
2016-01-05 19:37:08 +01:00
wm4 8fda7247ff ao_pulse: move format setting into a function
No real functional changes.
2016-01-05 19:34:34 +01:00
wm4 6147bcce35 audio: fix format function consistency issues
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".
2015-06-26 23:06:37 +02:00
Marcin Kurczewski f43017bfe9 Update license headers
Signed-off-by: wm4 <wm4@nowhere>
2015-04-13 12:10:01 +02:00
wm4 b7325b2f64 ao_pulse: drop video role; fixes random muting
The details are described in #1173.

This "features" causes problems to users so often, it's better to remove
it.

Fixes #1173.
2015-03-24 22:07:14 +01:00
wm4 e01750020d ao_pulse: listen for hotplug events
This requires jumping through multiple hoops on fire. Since the
PulseAudio API is virtually undocumented, I'm not sure if this is
correct either. We only react to sink events, and only to the NEW/REMOVE
events. CHANGE events are ignored, because PulseAudio fires them far too
often - even if the system is completely idle! If pa_sink_info.name can
change, we're in trouble. pa_sink_info.description is not so important,
but it'd also be a bit un-nice if it can change, and we don't update it.

The weird way how the actual AO and the hotplug context share the same
struct (ao) comes in handy here, although context_success_cb() still had
to be duplicated from success_cb() - the unused argument has a different
type.
2015-02-12 17:18:43 +01:00
Ben Boeckel b1d47786d8 ao_pulse: plug a memory leak 2015-01-25 01:26:11 +01:00
wm4 2c9180f47b ao_pulse: exit AO if stream fails
This can for example reproduced by killing the pulseaudio server. If
this happens, just try to reload the AO, instead of breaking everything
forever.
2015-01-11 04:19:40 +01:00
wm4 dc2d0539c7 ao_pulse: disable latency calculation hacks by default
This used to be required to workaround PulseAudio bugs. Even later, when
the bugs were (partially?) fixed in PulseAudio, I had the feeling the
hacks gave better behavior. On the other hand, I couldn't actually
reproduce any bad behavior without the hacks lately. On top of this, it
seems our hacks sometimes perform much worse than PulseAudio's native
implementation (see #1430).

So disable the hacks by default, but still leave the code and the option
in case it still helps somewhere. Also, being able to blame PulseAudio's
code by using its native API is much easier than trying to debug our own
(mplayer2-derived) hacks.
2015-01-07 22:23:38 +01:00
wm4 5db0fbd95e audio/out: consistently use double return type for get_delay
ao_get_delay() returns double, but the get_delay callback still
returned float.
2014-11-09 11:45:04 +01:00
wm4 b814b7ca84 audio: add --audio-client-name option
The main need I see for this is with libmpv - it would be confusing if
some application showed up as "mpv" on whateverthehell PulseAudio uses
it for (generally it does show up on various PA GUI tools).
2014-11-07 15:54:35 +01:00
wm4 f432a584b9 ao_pulse: implement AO device listing API
While conceptually this sink stuff in PulseAudio does just the right
thing, actually listing the sinks is unbelievable complicated. Not only
is the idea that listing them should happen asynchronously completely
bullshit (who the fuck runs the PulseAudio server on a separate
computer), but the way this is done is full of bullshit too. Why
separate callbacks for each device? Why this obtuse mainloop shit?
Especially the mainloop shit makes it actively worse than doing things
manually with pthread primitives, and the reason for that (different
mainloop implementations for GUIs?) is laughable too. It's like they
chose the most complicated API possible just because they attempted
to "abstract" basic mechanisms in order to handle "everything". While
I don't claim to design the best APIs, this API is fucking terrible
without any excuse. (End of rant.)
2014-10-10 18:42:43 +02:00
wm4 a25e936540 ao_pulse: move setup code to separate function
All the dumb crap in pa_init_boilerplate() is needed to talk to the
audio server at all. Might also fix some subtle bugs in the init code
(which is strange, because the original file was contributed by the
devil himself).
2014-10-10 18:42:06 +02:00
wm4 4b2f81a36f ao_pulse: don't use pa_format_info_to_sample_spec()
This function is available starting with PulseAudio 2.0, while we only
require 1.0. This broke compilation on Ubuntu 12.04.5 LTS.

Use our own function to calculate the buffer size, which is actually
simpler and needs slightly less code.

Hopefully fixes #1154.
CC: @mpv-player/stable
2014-10-06 21:49:26 +02:00
wm4 0d4e245de7 ao_pulse: change suspend circumvention logic
Commit 957097 attempted to use PA_STREAM_FAIL_ON_SUSPEND to make
ao_pulse exit if the stream was started suspended.

Unfortunately, PA_STREAM_FAIL_ON_SUSPEND is active even during playback.
If you pause mpv, pulseaudio will close the actual audio device after a
while (or something like this), and unpausing won't work. Instead, it
will spam "Entity killed" error messages.

Undo this change and check for suspended audio manually during init.

CC: @mpv-player/stable
2014-10-04 23:30:07 +02:00
wm4 9570976255 ao_pulse: refuse to start suspended
Sometimes, ao_pulse starts in suspended mode, which means playback is
essentially paused in pulseaudio. This gives the impression that mpv is
hanging, since it times video against the audio playback progress, and
audio never makes progress in this state.

I'm not sure if this will help - possibly it does with mixed
pulseaudio/alsa setups. However, if the alsa setup has the pulseaudio
plugin, alsa will hang too. But there's still a chance we get less
blame for pulseaudio messes.
2014-10-03 23:04:12 +02:00
wm4 c2fa9f6629 ao_pulse: digital pass-through
Should be able to pass-through AC3, DTS, and others.

It seems PulseAudio wants players to fallback to PCM on certain events
signaled by the server, but we don't implement that. There's not much
documentation available anyway.
2014-09-23 23:11:55 +02:00
wm4 7230d88c7e ao_pulse: correctly wait for stream state
This works similar to condition variables; for some reason this
apparently worked fine until now, but it breaks with passthrough mode.
2014-09-23 23:11:55 +02:00
wm4 601fb2f93a ao_pulse: use pa_stream_new_extended()
Needed for compressed audio pass-through later.
2014-09-23 23:11:55 +02:00
wm4 b745c2d005 audio: drop swapped-endian audio formats
Until now, the audio chain could handle both little endian and big
endian formats. This actually doesn't make much sense, since the audio
API and the HW will most likely prefer native formats. Or at the very
least, it should be trivial for audio drivers to do the byte swapping
themselves.

From now on, the audio chain contains native-endian formats only. All
AOs and some filters are adjusted. af_convertsignendian.c is now wrongly
named, but the filter name is adjusted. In some cases, the audio
infrastructure was reused on the demuxer side, but that is relatively
easy to rectify.

This is a quite intrusive and radical change. It's possible that it will
break some things (especially if they're obscure or not Linux), so watch
out for regressions. It's probably still better to do it the bulldozer
way, since slow transition and researching foreign platforms would take
a lot of time and effort.
2014-09-23 23:09:25 +02:00
wm4 f744aadb77 ao_pulse: dump library version etc.
Might help with debugging.

Unfortunately, there doesn't seem to be a way to get the actual
pulseaudio server version.
2014-09-10 23:14:06 +02:00
wm4 b578abe81b ao_pulse: fix typo in error message
Closes #1076.
2014-09-08 17:19:53 +02:00
wm4 439a05d8c3 audio/out: remove old things
Remove the unnecessary indirection through ao fields.

Also fix the inverted result of AOCONTROL_HAS_TEMP_VOLUME. Hopefully the
change is equivalent. But actually, it looks like the old code did it
wrong.
2014-09-06 02:30:57 +02:00
wm4 bc6359313f ao_pulse: allow disabling timing bug workarounds
Add an option that enables using native PulseAudio auto-updated timing
information, instead of the manual calculations added in mplayer2 times.
You can use --ao=pulse:no-latency-hacks to enable the new code. The code
is almost the same as the code that was removed with commit de435ed5,
but I didn't readd some bits I didn't understand. Likewise, the option
will disable the code added with that commit.

In my tests this seemed to work well, though the A/V sync display looks
funny when seeking.

The default is still the old behavior.

See issue #959.
2014-07-26 23:20:09 +02:00
wm4 77d9e4b8a9 ao_pulse: remove hacks for ancient PulseAudio versions
This was needed by very old (0.9) versions only. Get rid of it.

Unfortunately, I can't cross-check with the original bug report, since
the bug URL leads to this:

Internal Server Error

TracError: IOError: [Errno 2] No such file or directory: '/home/lennart/svn/trac/pulseaudio/VERSION'
2014-07-26 23:19:48 +02:00
wm4 80d36a0aa2 ao_pulse: fix potential compilation problem
It seems at least on some platforms (OSX 10.9), the POSIX wait()
function becomes visible, and conflicts with this unrelated function.
Just rename it.
2014-07-22 19:26:53 +02:00