Also clarify the statement about what we expect to happen by default.
It's well possible that distros at some point will fix their ALSA
configuration, and e.g. enable the upmix plugin by default.
This should work well with most audio APIs, except ALSA. A long-winded
explanation is provided how to make ALSA multichannel output work.
All other AOs should have no such problems. Of course it's possible
that previously unknown issues arise, because I assume that enabling
multichannel audio is actually relatively rare.
This also disables codec downmix by default, which could change the
audio output due to different mixing in the codec and libavresample.
Fixes#1313.
If no-block was given, the device would be opened with SND_PCM_NOBLOCK.
Also, after opening, blocking mode was unconditionally enabled anyway
with snd_pcm_nonblock(). Further, if opening with SND_PCM_NOBLOCK
failed, opening was retried without this flag.
This doesn't make any sense to me, and I've never heard of someone using
this suboption. I suspect it has to do with ancient ALSA bugs or API
caveats. Remove it and simplify the code.
Some ALSA plugins take non-interleaved audio, but treat it as
interleaved, which results in various funny bugs. Users keep hitting
this issue, and it just doesn't seem worth the trouble.
CC: @mpv-player/stable
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.
wm4
Stefano Pigozzi
Martin Herkt