| Commit message (Collapse) | Author | Age | Files | Lines |
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It existed for XP-compatibility only. There was also a time where
ao_wasapi caused issues, but we're relatively confident that ao_wasapi
works better or at least as good as ao_dsound on Windows Vista and
later.
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Same deal as with previous commit. "waveext" is less arbitrary and at
least supports 3/7 channels.
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ingore-chmap -> ignore-chmap
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Until now, this was for AC3 only. For PCM, we used AudioUnit in
ao_coreaudio, and the only reason ao_coreaudio_exclusive exists
is that there is no other way to passthrough AC3.
PCM support is actually rather simple. The most complicated
issue is that modern OS X versions actually do not support
copying through the data; instead everything must go through
float. So we have to deal with virtual and physical format
being different, which causes some complications.
This possibly also doesn't support some other things correctly.
For one, if the device allows non-interleaved output only, we
will probably fail. (I couldn't test it, so I don't even know
what is required. Supporting it would probably be rather
simple, and we already do it with AudioUnit.)
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ao_coreaudio uses AudioUnit - the OSX software mixer. In theory, it
supports multichannel audio just fine. But in practice, this might be
disabled by default, and the user is supposed to select a multichannel
base format in the "Audio MIDI Setup" utility.
This option attempts to change this setting automatically. Some possible
disadvantages and caveats are listed in the manpage additions. It is off
by default, since changing this might be rather bad behavior for a
normal application.
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The build failed because rst2pdf apparently has problems with
page breaks. In this case, the link to the ALSA upmix guide was
causing a page break in an admonition block. My guess is that
rst2pdf screws up when it can’t fill at least one line of text
following a page break, so I worked around this by making that
paragraph a little longer. Seems to do the trick.
I also shortened the URL using GitHub’s service because it was
causing some rather unsightly formatting in the manpage output.
Maybe we should just build HTML instead of a PDF.
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remove depricated and convoluted validation. refer instead to the
--audio-device option.
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This was requested, more or less.
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I'm not sure how common this behavior possibly is; well whatever. This
option will allow reproducing such behavior, and help debugging it.
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Pretty useful for debugging, although a bit useless or possibly
misleading too (see comments in the manpage).
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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.
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It's just completely useless. We have good native support for all 3
desktop platforms, and ao_sdl or ao_openal as fallbacks.
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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.
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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.
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Also, don't use the "hw" device, but "plughw". ALSA docs say never to
use "hw".
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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.
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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
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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.
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This additional sub-directory doesn't serve any purpose anymore. Get rid
of it.
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