This is just a conceptual issue, since for now every channel count has
an associated standard layout.
But should the max. channel count ever be bumped, some things would stop
function if mp_chmap_from_channels() refused to work for any channel
count within the allowed range.
In the AVFrame-style system (which we inreasingly map our internal data
stuctures on), buffers and plane pointers don't necessarily have a 1:1
correspondence. For example, a single buffer could cover 2 or more
planes, all while other planes are covered by a second buffer, and so
on. They don't need to be ordered in the same way.
Change mp_audio_get_allocated_size() to retrieve the maximum size all
planes provide. This also considers the case of planes not pointing to
buffer start.
Change mp_audio_realloc() to reset all planes, even if corresponding
buffers are not reallocated. (The caller has to be careful anyway if it
wants to be sure the contents are preserved on realloc calls.)
If you try to play surround with dmix, it will advertise surround and
lets you set more than 2 channels, but will report a stereo channel map,
with the extra channels identified as NA. We could handle this now, but
we don't want to (because it's excessively stupid).
Do it only if the channel map is not what we requested, instead of just
acting if it contains NA entries at all. This avoids that we hurt
ourselves in the unlikely but possible case we actually have to use
channel maps with NA entries.
So successful playback and user quit can be distinguished, for whatever
reason you may want to do this.
Normally, the "quit" command can be customized, but this does not work
for quit commands sent by the terminal signal handler. One solution
would be introducing something like "ON_SIGNAL" (equivalent to
"CLOSE_WIN"), but considering there are a bunch of possible signals, I'd
rather not get into this. So go with the dumb solution.
Probably fixes#2029.
gl_video_set_options() does not acquire ownership of the opts parameter
or its contents. In case of vo_cmdline, opts will point to temporary
memory. This memory will be free'd at a later point, and p->opts will
point to free'd memory on the next reinitialization.
The fix is pretty ugly, but it's a quick bug fix. This can probably be
removed once VO sub-options are exposed as properties.
Fixes#2035.
If the audio API takes a while for starting the audio callback, the
current heuristic can be off. In particular, with very short files, it
can happen that the audio callback is not called before playback is
stopped, so no audio is output at all.
Change draining so that it essentially waits for the ringbuffer to
empty. The assumption is that once the audio API has read the data
via the callback, it will always output it, even if the audio API
is stopped right after the callback has returned.
If a frame could only be partially filled with real audio data, the
silence wasn't written at the correct offset. It could have happened
that the remainder of the frame contained garbage.
(This didn't happen in the more common case of playing dummy silence.)
Used a wrong condition, and I suppose it could crash in some situations.
Change it to lazily initialize the hotplug stuff, like the
audio-device-list property does.
This was matching e.g. both "foo/bar" and "foobar" against "foo", when
only the former should match. This could cause more property
notifications than necessary.
Absence of license header implies LGPL, as mentioned in the "Copyright"
file. But vaapi.h contains some code taken from the mplayer-vaapi
patch, which was under the typical MPlayer license.
All vo_gl.c related code has been GPL+LGPL dual-licensed. The OSD code
is no exception and is also derived from vo_gl.c. Thus it should have
the same license (although I think technically speaking sub-licensing
it by removing one of the licenses is ok).
Commits 92b27be and f4ce99d removed high-fps logic to to a bug. That bug was
a missing parenthesis around everything after duration >= 0 && ... at the
removed code.
This patch restores the removed code, fixes the bug and then refactors the
code a bit.
This reverts commit f1746741de.
Together with the other revert, this fixes#2023 (the reason being
broken framedrop handling - it was dropping frames when it shouldn't).
read_output_surface() could fail and return NULL.
Also, make sure we don't set the image to a size larger than the
allocated size. Normally this shouldn't happen, but in theory it could
in corner cases; this is for robustness.
This provides a new method for enabling spdif passthrough. The old
method via --ad (--ad=spdif:ac3 etc.) is deprecated. The deprecated
method will probably stop working at some point.
This also supports PCM fallback. One caveat is that it will lose at
least 1 audio packet in doing so. (I don't care enough to prevent this.)
(This is named after the old S/PDIF connector, because it uses the same
underlying technology as far as the higher level protoco is concerned.
Also, the user should be renamed that passthrough is backwards.)
This deprecates the --ad-spdif-dtshd option, and replaces it with a
pseudo decoder. This means ad_spdif will report two decoders, "dts" and
"dts-hd", of which the second simply enables what the option did.
The --ad-spdif-dtshd option will actually be deprecated in the next
commit.
This makes no sense, because the format can't be converted anyway. It
just sets up the filter chain init code, which will vomit a bunch of
useless and confusing messages. So uninit and fail explicitly when this
happens.
We can't do much in this case, but at least we can not call the vdpau
API functions with too large sizes. Apparently the API considers this
undefined behavior, and random stuff might happen.
The previous code was not wrong, but I'd claim this makes the code more
robust. If a situation could happen in which the passed surface size is
incorrect, we could have passed a too small image, and
VdpOutputSurfaceGetBitsNative could have randomly overwritten memory.
Check the maximum size of video surfaces, and refuse initialization if
the video is too large for them.
Maybe we could do something more sophisticated, like inserting a
software scaler. On the other hand, this would have a very questionable
benefit, as it would be guaranteed to be too slow.
When starting in paused mode, no audio is written to the device at all,
because writing audio implicitly unpauses the AO. If the file is very
small, and all audio fits within the AO buffer, this accidentally
triggered the EOF condition. (In unpaused mode, it would write all
audio, end playback, and then wait until the AO has everything played.)
This is better, because now we call swr_get_delay() with the output
samplerate, instead of with the input samplerate and then multiplying it
with the ratio and rounding it up.
Apparently, just running ./waf and hoping that it will be run with a
Python interpreter doesn't necessarily work. The workaround is pretty
simple and reliable.
The "osd" command cycles between 4 states (OSD level 0-3), which is
probably confusing and inconvenient. OSD levels 0 and 2 are rarely
needed. I would claim there is normally not much of a need to completely
disable OSD by setting level 0 during playback. Level 2 is just slightly
less information than level 3, and I'm not sure why it exists at all.
Change it so that it toggles between level 3 and 1. Note that this
ignores the default OSD level. If the default is 3, the first use of
this key will set it to 3 again. Just assume 1 is the default. If
someone complains, this could be improved.
There's a short time during loading where external commands can add
external streams even before the main file is loaded (like during ytdl
hook execution). The track list is printed every time an external track
is added via commands. This was quite awkward when ytdl was adding
multiple streams, so don't print it in this stage. They are printed
anyway at the end of the loading process.
wm4
torque
Avi Halachmi (:avih)
Jaime Marquínez Ferrándiz
Niklas Haas