Normally, PulseAudio accepts any combination of sample format, sample
rate, channel count/map. Sometimes it does not. For example, the channel
rate or channel count have fixed maximum values. We should not fail
fatally in such cases, but attempt to fall back to a working format.
We could just send pass an "unset" format to Pulse, but this is not too
attractive. Pulse could use a format which we do not support, and also
doing so much for an obscure corner case is not reasonable. So just pick
a format that is very likely supported.
This still could fail at runtime (the stream could fail instead of going
to the ready state), but this sounds also too complicated. In
particular, it doesn't look like pulse will tell us the cause of the
stream failure. (Or maybe it does - but I didn't find anything.)
Last but not least, our fallback could be less dumb, and e.g. try to fix
only one of samplerate or channel count first to reduce the loss, but
this is also not particularly worthy the effort.
Fixes#2654.
Apparently, the firmware will ignore pixel_x/pixel_y if the numeric
value of them gets too high (even if they indicate square pixel aspect
ratio). Even worse, the destination rectangle is ignored completely,
and the video frame is simply stretched to the screen. I suspect this
is an overflow or weird sanity check within the firmware.
Work it around by limiting the fields to 16000, which is an arbitrary
but apparently working limit.
There are a lot of incorrectly encoded subtitles with .ass extension
and non-ass subtitles (srt, ssa) with such extension, so we need to
try codepage detection even for .ass.
Signed-off-by: wm4 <wm4@nowhere>
Given 5.1(side), this lets it pick 5.1 from [5.1, 7.1]. Which was
probably the original intention of this replacement stuff. Until now,
the opposite was done in some cases.
Keep the old heuristic if the replacement is not perfect. This would
mean that a subset of the channel layout is an inexact equivalent, but
not all of it.
(My conclusion is that audio output APIs should be designed to simply
take any channel layout, like the PulseAudio API does.)
Unify and clean up listing and selection. Use common enumerator code for both
operations to avoid duplication or inconsistencies.
Maintain, but significatnly simplify manual device selection by id, name or
number. This actually fixes loading by name which didn't really work before
since the "name" displayed by --audio-device=help differed from that used to
match the selection, which used the device "description" instead.
Save the selected deviceID in the private structure for later loading. This will
permit moving the device selection into the main thread in a future commit.
GLSL in GLES 2.0 did not have line continuation in its preprocessor.
This broke shader compilation. It also broke subtitle rendering in
vo_rpi, which reuses some of the OpenGL code.
Line continuation was finally added in GLES 3.0, which is perhaps the
reason why ANGLE accepted it.
When using --start with timeline/ordered chapters, then the
timeline_switch_to_time() function will look at playback_initialized
whether to rselect the currently selected streams on the demuxer level.
So we need to set this field to true at an earlier stage during
initialization, and in particular before the code for --start is called.
Apparently it's only wine where the qpc_position returned by
IAudioClock_GetPosition can be overflowed. So actually do the rescaling
correctly, but throw away the result if it looks unreasonable.
this fixes a regression in 5afa68835a
In particular, the player inserts af_scaletempo by default now when
changing speed, so the suggestion to add this was missleading and
counter-productive.
Slightly change how it is decided when a new packet should be read.
Switch to demux_read_packet_async(), and let the player "wait properly"
until required subtitle packets arrive, instead of blocking everything.
Move distinguishing the cases of passive and active reading into the
demuxer, where it belongs.
Just so I can remove a few lines from dec_sub.c.
This is slightly inelegant, as the whole subtitle file has to be read
into memory, converted at once in memory, and then provided to
libavformat in an awkward way by creating a memory stream instead of
using demuxer->stream. It also won't be possible to force the charset on
subtitles in binary container formats - but this wasn't exposed before,
and we just hope this won't be ever needed. (One motivation was fixing
broken files with non-UTF8 muxed.) It also won't be possible to change
the charset on the fly, but this was not exposed either.
Just simplify by removing parts not needed anymore. This includes
merging dec_sub allocation and initialization (since things making
initialization complicated were removed), or format support queries (it
simply tries to create a decoder, and if that fails, tries the next
one).
So that the video FPs is not required at initialization, and can be set
later.
(As for whether this MicroDVD crap is worth the trouble to handle it
"correctly": MicroDVD files are unfortunately still around, and in at
least one case using the video FPS seemed to help indeed.)
Always preroll by default if the cue (index) information indicates
overlapping subtitles.
Increase the amount of maximum data it will skip to get such subtitles
to 10 seconds. Since the index information can reliably tell whether
reading earlier is needed, the maximum should be rarely actually used,
thus we can set it high. On the other hand, the "old" prerolling
mechanism always has to skip the maximum amount of data; thus the method
using the index gets its own option to control the maximum amount of
data to skip.
(As more and more files With newer mkvtoolnix versions are muxed, and
with this new and hopefully sane default established, these options can
probably be removed in the future.)
Untested, but should be fine. Broken by commit 0a0bb905.
Also fix the include statement in context_rpi.c, which caused another
compilation failure. Also untested. (Because I'm lazy.)
Fixes#2638.
Keeping ASS_Renderers around for a potentially large number of subtitle
tracks could lead to excessive memory usage, especially since the libass
cache is broken (caches even unneeded data), and might consume up to
~500MB of memory for no reason.
Kevin Mitchell