I'd like some names to be more descriptive, but to work with 15 chars
limit we have to make some sacrifice.
Also because of the limit, remove the `mpv/` prefix and prioritize
actuall thread name.
It was found that this causes issues with at least ao_coreaudio,
essentially revealing a way bigger issue:
Some AOs don't check for 0 and/or have no way to deal with short writes.
Someone will have to figure out a fix later but get rid of the direct
cause for now.
This reverts commit ae908a70ce.
ao_read_data() is used by pull AOs potentially from threads managed by
external libraries. These threads can be sensitive to blocking.
For example the pipewire ao is using a realtime thread for the
callbacks.
since i was going to fix the include order of stdatomic, might as well
sort the surrouding includes in accordance with the project's coding
style.
some headers can sometime require specific include order. standard
library headers usually don't. but mpv might "hack into" the standard
headers (e.g pthreads) so that complicates things a bit more.
hopefully nothing breaks. if it does, the style guide is to blame.
replace it with <stdatomic.h> and replace the mp_atomic_* typedefs with
explicit _Atomic qualified types.
also add missing config.h includes on some files.
Pull AOs work off of a callback that relies on mpv's internal timer. So
like with the related video changes, convert all of these to nanoseconds
instead. In many cases, the underlying audio API does actually provide
nanosecond resolution as well.
There's a lot of wild 1e6, 1000, etc. lying around in the code. A macro
is much easier to read and understand at a glance. Add some helpers for
this. We don't need to convert everything now but there's some simple
things that can be done so they are included in this commit.
Linux and macOS already use nanosecond resolution for their sleep
functions. It was just being converted from microseconds before. Since
we have mp_time_ns now, go ahead and bump the precision here. The timer
for windows uses some timeBeginPeriod thing which I'm not sure what it
does really but whatever just convert the units to ms like they were
doing before. There's really no reason to keep the mp_sleep_us helper
around. A multiplication by 1000 is trivial and underlying OS clocks
have nanosecond precision.
This is the most supported in standard layout, if we request more it
tends to fallback to stereo instead. Also channels mask is 32-bit and it
can get truncated.
A bit different from the OPT_REPLACED/OPT_REMOVED ones in that the
options still possibly do something but they have a deprecation
message. Most of these are old and have no real usage. The only
potentially controversial ones are the removal of --oaffset and
--ovoffset which were deprecated years ago and seemingly have no real
replacement. There's a cryptic message about --audio-delay but who
knows. The less encoding mode code we have, the better so just chuck
it.
The idea behind period_size is that it's the minimum amount of data
that your audio subsystem wants to fetch.
For PulseAudio, this is given by the minreq buffer attribute, which
is in bytes for all channels. Hence the divisions.
This change sets the device_buffer member of the ao struct for
the JACK ao to whatever value we read during init.
While JACK allows changing the audio buffer size on-the-fly
(you can do this for example through DBus), having it somehow
reconfigure the entire audio buffer logic of mpv that depends
on device_buffer in some way when that happens only leads to
audio dropout and weird code. device_buffer's role is mostly for
prebuffer from what I understand, which means that simply setting
it to its current value in the init function is fine.
In addition to the patch, appropriate additions to the mpv.conf file
will of course be needed for this to work. For example on my system:
audio-device=oss//dev/dsp4
audio-spdif=ac3,dts,dts-hd,eac3,truehd
This has been tested using recent FreeBSD-13.1-stable, using external
surround processors (both a Trinnov Altitude 16 and an LG OLED that
supports Dolby Atmos, and connected with HDMI to an NVidia RTX 2070).
Author and tester: David G Lawrence <dg1007@dglawrence.com>
There's an edge cause with gapless audio and pausing. Since, gapless
audio works by sending an EOF immediately, it's possible to pause on the
next file before audio actually finishes playing and thus the sound gets
cut off. The fix is to simply just always do an ao_drain if the ao is
about to set a pause on EOF and we still have audio playing.
Fixes#8898.
The difference this makes is that the OS API will notice
when we underrun (as opposed to being fed silence).
Other AOs mostly seem to not do this because they've committed
to filling a buffer of a certain size no matter what, but I have
not observed any ill effects for AudioTrack in my testing.
This looks like a pretty bad bug but only became a problem
with the last commit that allows rates like 22.5kHz to pass through
directly instead of being resampled.
wireplumber uses the media role when the node is first created.
To have the property available at this point reliably we need to set it
directly when creating the stream/node.
As the role property is interpreted by wireplumber it can only be
evaluated when creating the stream. The current, dynamic mechanism is
racy so revert it.
See: #11253Fixes: #12041
This reverts commit 535cd6f313.
Doing a pw_thread_loop_wait() without checking conditions is invalid.
The thread loop could be signalled for other reasons and in this case
the wait needs to continue.
PipeWire added such additional signaling in
commit 33be898130f0 ("thread-loop: signal when started").
This meant that for_each_sink would return before the callbacks have
fired and session_has_sink() would incorrectly return "false", failing
the initialization of ao_pipewire.
Fixes#11995
Now that Debian 12 is release bump the minium required version to what
is provided in Ubuntu Jammy (22.04).
The same as has been done for the wayland dependencies.
Signed-off-by: Thomas Weißschuh <thomas@t-8ch.de>
Instead of brute forcing the name until it is set, without any error
checking and expecting it would start to work, fallback to client name
if initial request fails.
Fixes player going into infinite loop with very long title names. The
API rejects unreasonably long names, which make sense.
As for alleged "weird race condition in the IAudioSessionControl itself"
I cannot comment. It works on my end and even if it fails, it is not a
critical error or even something that we should care about... and
obviously not hang the whole player for that.
Fixes: #11803
By making the data thread realtime it is able to serve requests faster
and more reliable reducing crackling in certain situations.
As the mpv callbacks that are running on the data thread are all
non-blocking and very short this should be safe.
The same mechanism is also used by pw-cat and the alsa plugin shipped by
pipewire.
c784820454 introduced a bool option type
as a replacement for the flag type, but didn't actually transition and
remove the flag type because it would have been too much mundane work.
