This changes the code so that it does the same as MPlayer, mplayer2
and mpv before ref-counted AVFrame. The problem is that get_buffer2
is called with aligned frame dimensions, while get_buffer didn't. This
breaks the mpv video frame size change detection.
Followup to 8df7127. This refines the condition for front ordering the
condition to account for minimized or hidden state where the window should go
to the front only as a consequnce of user interaction.
FFmpeg (as well as Libav) have two layouts called "6.1":
AV_CH_LAYOUT_6POINT1 and AV_CH_LAYOUT_6POINT1_BACK. We call them "6.1"
and "6.1(back)". Change the default layout for 7 channels as well to
return the same layout as av_get_default_channel_layout(). (Looks a bit
questionable, but for now it's better to follow FFmpeg.)
When using --fs `vo_cocoa_fullscreen` was called from the primary thread. This
occurred inside `vo_cocoa_config_window` which is scheduled for excution on the
primary thread with libdispatch).
All of this caused spam from NSRecursiveLock in standard output.
When going in and going out of full screen the player lost information on the
movableByWindowBackground behaviour. There were some hacks in place to fix it
but they were broken with the recent native fullscreen changes in 74c15ec6.
This commit removes the problem at the root and removes the hacks. The delegate
method `isMovableByWindowBackground` seems to be called after setFrame and
setPresentationOptions so change fs state in opts before that.
Fixes#83
It turns out that ALSA's 4 channel layout is different from mpv's and
ffmpeg's 4.0 layout. Thus trying to do 4 channel output led to incorrect
remixing via lib{av,sw}resample.
Fix the default layouts for the internal filter chain as well, although
I'm not sure if it matters at all.
The libavresample version of the current Libav stable release lacks the
avresample_set_channel_mapping() function. (FFmpeg's libswresample seems
to be fine, because they added swr_set_channel_mapping() first.)
Add a cheap/slow workaround to do channel reordering on our own. We
don't use the recently removed MPlayer code (see commit 586b75a),
because that is not generic enough.
The functionality should be the same as with full-featured
libavresample, and any differences are bugs. It's probably slower,
though.
Apparently useful for dumping DVD. Could also be used to rip streams
with libquvi and such, but for that there are better tools. Actually
I doubt there aren't better tools to dump DVDs, but whatever, this was
a feature request, so I don't need a good reason.
af_reinit() is responsible for inserting automatic conversion filters
for channel remixing, format conversion, and resampling. We don't
require that a single filter can do all these (even though
af_lavrresample does nearly all of this, sometimes af_format has to be
used instead for format conversions). This makes setting up the chain
more complicated, and a way is needed to prevent endless appending of
conversion filters if a conversion is not possible.
Until now, this used a stupidly simple yet robust static retry limit to
detect failure. This is perfectly fine, and the limit (20) was good
enough to handle about ~5 filters. But with more filters, and if each
filter requires 3 additional conversion filters, this would fail. So
raise the limit to 4 retries per filter. This is still stupidly simple
and robust, but won't arbitrarily fail if the filter count is too large.
To make this easier, get rid of the direct mapping of the
AF_FORMAT_BITS_MASK bit field to number of bytes. This way we can throw
away the unused AF_FORMAT_48BIT and don't have to add ..._56BIT.
The snd_pcm_hw_params_test_format() call actually crashes in alsa-lib if
called with SND_PCM_FORMAT_UNKNOWN, so the already existing fallback
code won't work in this case.
Make all AOs use what has been introduced in the previous commit.
Note that even AOs which can handle all possible layouts (like ao_null)
use the new functions. This might be important if in the future
ao_select_champ() possibly honors global user options about downmixing
and so on.
The point is selecting a minimal fallback. The AOs will call this
through the AO API, so it will be possible to add options affecting
the general channel layout selection.
It provides the following mechanism to AOs:
- forcing the correct channel order
- downmixing to stereo if no layout is available
- allow 5.1 <-> 5.1(side) fallback
- handling "unknown" channel layouts
This is quite weak and lots of code/complexity for little gain. All AOs
already made sure the channel order was correct, and the fallback is of
little value, and could perhaps be done in the frontend instead, like
stereo downmixing with --channels=2 is handled. But I'm not really sure
how this stuff should _really_ work, and the new code will hopefully
provides enough flexibility to make radical changes to channel layout
negotiation easier.
If one of the input or output is an unknown layout, but the other is
known, it can still happen that channels are remixed randomly. Avoid
this by forcing default layouts in this case. (Doesn't work if the
channel counts are different.)
Now mpv's channel map is used to map each channel to a speaker. This
allows in theory for playback of any layout for which ao_openal
actually has a speaker defined. Also add the back-center (BC) speaker,
which allows playback of 6.0 audio. Enabling more layouts by adding
other speakers would be possible, but I'm not sure about the speaker
positions.
This allows supporting 5 channel audio (which can be eother 5.0 or 4.1).
Fallback doesn't work yet. It will do nonsense if the channel layout
doesn't match perfectly, even though it's similar.
Add a CHECK_ALSA_ERROR macro to report ALSA errors. This is similar to
what vo_vdpau does. This removes lots of boiler plate, it almost gives
me the feeling the ao_alsa initialization code is now readable. This
change is squashed with the reformatting, because both changes are
just as noisy and useless.
Using demux_rawaudio and the --rawaudio-channels option is useful for
testing channel map stuff. The libavcodec PCM decoder normalizes the
channel map to ffmpeg order, though. Prevent this by forcing the
original channel map when using the mp-pcm pseudo decoder entry (used by
demux_rawaudio and stream/tv.c only).
Like most other AOs, ao_pulse set the channel count only, always using a
default layout. Try to set the exact layout.
For this, we need a big lookup table to map waveex/lavc/mpv speaker
position to PulseAudio's, since PA_CHANNEL_POSITION_ is apparently not
compatible to waveext, and I haven't seen any API functions that would
help mapping them.
Completely untested. (Let's leave that to someone else...)
This helps passing the channel layout correctly from decoder to audio
filter chain. (Because that part "reuses" the demuxer level codec
parameters, which is very disgusting.)
Note that ffmpeg stuff already passed the channel layout via
mp_copy_lav_codec_headers(). So other than easier dealing with the
demuxer/decoder parameters mess, there's no real advantage to doing
this.
Make the --channels option accept a channel map. Since simple numbers
map to standard layouts with the given number of channels, this is
downwards compatible. Likewise for demux_rawaudio.
This is done in af_lavrresample now, and as part of format negotiation.
Also remove the remaining reorder_channel calls. They were redundant
and did nothing.
This actually breaks audio for 5/6/8 channels. There's no reordering
done yet. The actual reordering will be done inside of af_lavrresample
and has to be made part of the format negotiation.
