This is the new FFmpeg channel layout structure, which now
combines channel count and layout into a single location.
Only unspecified (channel count only) and native (channel layout
mask based) layouts are currently supported for the initial move
towards non-deprecated APIs.
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>
Simply returning out of this function leaks avpkt, need to always "goto
done".
Rewrite the logic a bit to make it more clear what's going on (IMO).
Fixes#9593
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
This brings my scaletempo2 benchmark down from ~22s to ~7s on my machine
(-march=native), and down to ~11s with a generic compile.
Guarded behind an appropriate #ifdef to avoid being ableist against
people who have the clinical need to run obscure platforms.
Closes#8848
This fixes audio encoding crashing under ASan.
When extended_data != data, FFmpeg copies more pointers from
extended_data (= the number of channels) than there really
are for non-planar formats (= exactly 1), but that's not our fault.
Regardless, this commit makes it work in all common cases.
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.)
