Caps returned from gst_pad_peer_get_caps_reffed () may not be writable.
If they are not is should cause an assertion in gst_caps_merge (),
however, sometimes assertions are disabled in binary builds of -base and
it's safer to just be sure the caps are writable. Also, check that the
reffed caps pointer is not NULL.
The length check isn't sufficient, an source might
report the correct length, but then still fail to
read the requested number of bytes for some reason.
https://bugzilla.gnome.org/show_bug.cgi?id=652642
This is especially needed when switching between a non-sparse and sparse
video stream, see bug #537382. It also lowers the time needed for switching
between streams a bit.
Previously we checked mute_csource to determine wheter we need to premultiply
volumes and mute values. That fails as we unrefs mute_csource and set it to
NULL after. Use an extra flag instead.
make_lossless_changes() returns the same structure that we're passing (probably
to enable chaining). Instead of reusing s and making it point to s2 as well,
keep using s2. Drop the assignment which in the 2nd case is a dead one anyway.
Autoplug formatters for streams if a formatter with secondary or
higher rank is found. Formatters are autoplugged when there is no
muxer or when the muxer doesn't implement the tagsetter interface.
Currently only the first formatter found is plugged, this might
help in lots of cases, but it doesn't solve the
'lamemp3 ! xingmux ! id3mux'
case.
https://bugzilla.gnome.org/show_bug.cgi?id=649841
In particular, in audio only cases whose (estimated) metadata provides bitrate
information, the buffer-size based on such bitrate (and buffer-duration)
will be much more reasonable than queue2 default buffer-size.
For streams at low bitrates we need to set a limit in time because the limit
in bytes might not reached too late, sometimes more than 30 seconds.
This limit can only be set if upstream is seekable (see #584104)
Closes#647769
These reconfigure based on the caps and plugin in converters if
necessary. This also makes switching between compressed and raw
streams work flawlessly without loosing the states of any element
somewhere or having running time problems.
Before playbin2 would use different selectors for raw audio and
compressed audio (and the same for video) and used different
pads from playsink. This made the involved logic much more
complex and was not implemented completely in playsink, which
made it impossible to support files with a compressed and
uncompressed stream that is support by the sink.
playbin2 handles raw/non-raw streams the same now and the
decision is left to playsink, which now can also handle
caps changes from raw to non-raw and the other way around.
Fixes bug #632788.
Fixes#648548. Orc generates bad code for
gst_videoscale_orc_resample_merge_bilinear_u32, so we'll use the
slightly slower two-stage process. I'd fix Orc, but it's hard to
get excited about fixing a feature that I'm planning to deprecate
and replace.
Remove the android/ top dir
Fixe the Makefile.am to be androgenized
To build gstreamer for android we are now using androgenizer which generates the
needed Android.mk files.
Androgenizer can be found here:
http://git.collabora.co.uk/?p=user/derek/androgenizer.git
